Computers
Most of people that work with computers don't know how it works. It Just works!
Infact computers were them most developed object last century, and they were the object that helped the develop of comunications, equipment and even it's selve.
Computers were the invention of the millenium they were made to help the ones that made it!

If you know how a computer works, you have an future!

What they are and how they work!
How does a computer work?
How....?
You may know how a cd-rom works, but you don't know a processor works.
That's what i am going to.....

First you need to learn ASM

ASM

ASM stands for assembly, witch is the cpu internal language.
To learn about ASM there is nothing better than the links i provide in the link selection
The best link for learning ASM is This one
I will continue later !

Music

Basic Knowledge:

OVERVIEW OF DIGITAL SOUND
If you are new to digital sound editing, it will be well worth your time to become familiar with some of the basic concepts. In this section we cover the most important fundamentals. However, we strongly recommend that you page through a book on digital audio and sound recording if you want to get the most out of  editing and digital signal processing features.
SOUND WAVES
You can think of air pressure as the density of air molecules. When an object vibrates or moves, it displaces air molecules causing a pressure change. This in turn, causes other air molecules to move. We don't hear air pressure changes caused by the weather. Instead, we hear air pressure differences that vary rapidly over time. When you hear a sound, you are sensing changes in the air pressure around your eardrum. These vibrations are then picked up by your ears and converted to electrical signals that your brain interprets as sound. If we were to graph the air pressure at your eardrum as a function of time while you were listening to a short sound, it might look like the waveform that follows.
LOUDNESS AND PITCH
When there is no sound wave, the air pressure is constant. This is perceived as silence. As the sound wave reaches your eardrum, the air pressure changes above and below the normal atmospheric pressure. The amount of change is perceived as the loudness of the sound. The loudness of a sound, called its amplitude, is usually measured as a fraction of a standard level, often in decibels (dB). The rate at which the air pressure changes is perceived as the pitch. In scientific terms, this term corresponds to the frequency of the wave. The frequency is usually measured in Hertz (Hz), or cycles per second. Sounds in nature are not as simple as the sine wave we graphed above. In reality, a sound would look something like the one drawn below. This irregular waveform does not have a periodic amplitude or frequency. 
TIMBRE
Complex waveforms like the one shown above are constructed by combining a number of simple waveforms (like the one in the first drawing) of different amplitudes and frequencies. This is why we perceive both high and low pitched sounds at once when we hear most natural sounds. The characteristic sound of a waveform (be it produced by a grand piano or a violin) is called its timbre. Timbre, also referred to as tone color, is said to be rich or full when there are many different frequencies in a sound. A sound from
a sine wave is considered dull by most people since it has only one frequency. The different frequencies in a sound, combined with the varying amplitudes of each frequency, make up the spectral content of a waveform. The spectral content, which you might say is the more scientific term for timbre, usually varies over time. Otherwise, the sound remains static and again sounds dull. The spectral characteristic of a waveform over time is the signature of a tone that allows you to describe it as string-like or horn-like.
ANALOG RECORDING AND PLAYBACK
Let's say you're recording with a microphone. As you hold the microphone up in the air and scream, the microphone converts the changes in air pressure into changes in electrical voltage. This is called an analog signal. If you were to graph the changing voltage inside a microphone cord, it would look exactly like the graph of the air pressure going up and down. To record your scream, you would send the signal to a medium such as magnetic tape which can store a replicate of the analog signal. To playback your recording, you need something to create the differences in air pressure that our ear interprets as sound, i.e. an audio speaker. Speakers operate by moving a cone from one position to another in a consistent manner. In order to move the cone either forward or backward the speaker must be driven by an electrical current. During playback, a tape or record player generates a current that is then fed to an amplifier. When connected to a speaker, the current moves the speaker in a way that reproduces the pressure changes sensed by the microphone during recording. Until recently, sound was always recorded as an analog signal on magnetic tape or vinyl grooves. One problem with storing a signal in this form is that it is hard to accurately record the analog signal without adding noise. When you make copies of your recording you have to convert it to an electrical signal and then re-record it, adding even more noise. Listen to a third generation cassette recording and you'll know what we're talking about. Also, editing with tape is a not an easy task, since you must always be fast-forwarding or rewinding to a section, splicing, etc. Tape-based editing is called linear editing.
DIGITAL RECORDING
With recent advances in computer technology, it has become efficient and economical to record sound waves using a process called digital sampling. In digital sampling, the analog signal of the sound wave is divided and stored as numbers that represent the amplitude of the wave over very small segments of time. For a moment, let's take a look at another process that is very similar to the way our computer makes sound – the making of movies and television. Given a scene with a person walking, we can slow down the speed at which the pictures are shown and see that each movement is captured by a different picture. As we speed up the rate of the pictures passing by, the motion becomes more fluid and eventually we stop noticing each individual picture. If we keep speeding up the movie the person appears to walk faster and faster and eventually ends up looking quite humorous. The important point is that a movie is just a collection of individual pictures. So how do people record a movie? Obviously they pick up a movie camera and film the action. The movie camera takes a series of pictures at a fast rate and saves them on the film. The movies we see are simply a collection of pictures that are played back in rapid succession. When we record a sound through the sound card in our computer (also called digitizing or sampling) we do much the same thing as the movie camera. The computer rapidly checks what position the microphone is in and saves it in the computer. When done recording, the computer has a collection of individual positions (normally called samples) which it can use to recreate the sound we have recorded. Individually, the samples are almost meaningless, much like an individual picture in a movie, but together they make up the recorded sound. This method of recording and playing sound is known as digitized sound. 
THE PC SPEAKER
A simple model for a digital system is one where a speaker cone can be in one of two positions, either in or out, corresponding to the numbers 1 and 0 stored in a computer. The normal position, in, is when the speaker is sitting at rest with no current applied. The speaker isn't moving and we don't hear anything. When current is applied the speaker cone moves to the other position, out. As the speaker moves to the out position it forces the air around it to move and we hear a small click or pop. If we leave the speaker sitting in the out position we again hear nothing since the speaker only produces sound when it is moving, not when it is stationary. Now if we move back and forth between the in and out positions we will hear a tone. As we move it faster and faster between the two positions (increasing the frequency) we will hear the tone increase in pitch. This model of a simple digital speaker system is exactly how the speaker inside your PC operates. Programs can put the speaker in either the in or out position to make sounds. Although there are more complicated methods to allow this type of system to produce sounds other than tones, they are beyond the scope of this introductory text.
EXPANDING THE MODEL
Instead of using just the two-position model with the in and out positions, let's suppose we had a system that allowed us to have 100 positions or even more. If this were the case then we would be able to make much more complex sounds. We could move it just a little bit or we could move it all the way out. This would allow us to have more precise control of the amplitude of the waveform. The more positions we have, the more flexibility we have in producing sound. For example, if we were to represent amplitude as a number from one to four, any values that fell between would be rounded to the closest value. This rounding error is called quantization noise. When more positions are available, rounding errors become smaller. You will often see a sound card referred to as 8-bit, or 16-bit. We can directly relate this to the number of positions in which we can place the speaker. With an 8-bit card we can place it in 256 different positions and with a 16-bit card we can place it in any of 65,536 positions. Although you might think that a 16-bit card should have twice as many positions as an 8-bit card, this is not the case. It actually has 256 times as many positions. Even though 16-bit samples take up twice as much space as 8-bit samples, it is recommended that when at all possible you use 16-bit samples to minimize quantization noise.
SAMPLING RATE
The number of times a sound waveform is checked for position each second is the sampling rate. The sampling rate is similar to the frame rate in movies. As you can imagine, with higher sampling rates you store more information about the sound's changing amplitude. This gives you more fidelity. As a matter of fact, it is impossible to accurately record frequencies above one-half of the sampling rate. This threshold frequency is called the Nyquist Frequency, and should be considered when selecting a sampling rate. Frequencies higher than the Nyquist Frequency show up as alias noise. The downside to very high sampling rates is that since each sample takes up space in memory (1 byte for 8-bit samples, 2 bytes for 16-bit samples), higher sampling rates will fill up your hard drive faster than lower sampling rates. For instance, a stereo digitized sound of 44,100 Hz 16-bit data (approximately what your CD player uses) lasting 10 seconds takes up almost 2 megabytes of space! This means if you have a 40 megabyte hard drive you couldn't even store 4 minutes of sound data, and that's without having any programs or other data on your system.
ADVANTAGES OF DIGITAL EDITING
The advantages of digital editing far outweigh the enormous storage requirements. Once you've recorded a sound as a digital sample on your hard drive, you have the ability to perform edits like copying, cutting, and pasting without losing any fidelity and, as some people like to brag, with accuracies of up to 0.000023 seconds (single sample spacing at 44,100 Hz sampling rate). With a visual editor  you can actually see a representation of the waveform to navigate through the sound file quickly and accurately. Another advantage of storing sound digitally is the availability of digital signal processing (DSP) techniques. Digital signal processing techniques can be used for filtering, simulating room acoustics, and other special effects to restore or enhance the original recorded sound. Finally, with a tool like Sound Forge, you can open and save your sound files to and from a number of different computer platforms, sound cards, and external samplers.
DIGITAL LEVELS
When recording to an analog medium such as magnetic tape, recording engineers always try to keep their meters as close to 0 VU (stands for Volume Unit, which is based on electrical currents) as possible. This ensures a high signal-to-noise ratio while preserving enough headroom to keep the tape from saturating and distorting. Recording a few peaks that go above 0 usually doesn't cause any problems since the tape saturation point is not an absolute. In the digital realm, where amplitudes are stored as discrete numbers instead of continuous variables, things are quite different. Instead of having a flexible and forgiving recording ceiling, we have absolute maximum amplitudes, -32,768 and 32,767, in 16-bit audio. No stored signal can ever have a value above these numbers. Everything beyond gets clamped to these values, literally clipping off the wave peaks. This chopping effect can add large amounts of audible distortion. If the clipping is very short and infrequent such as during a very loud snare hit, it can go unnoticed. But in general, it is safe to say that digital audio has absolutely no headroom. At what level, then, should a signal be recorded digitally? The standard method for digital metering is to use the maximum possible sample amplitude as a reference point. This value (32,768) is referred to as 0 decibels, or 0 dB. Decibels are used to represent fractions logarithmically. In this case, the fraction is: sample amplitude divided by the maximum possible amplitude. The actual equation used to convert to decibels is: dB = 20 log (amplitude/32,768) Say you have a sine wave with a peak amplitude of 50% of full scale. Plugging the numbers in gives you 20 log (0.50) = -6.0 dB. In fact, every time you divide a signal's amplitude by two, you subtract its dB value by 6 dB. Likewise, doubling the amplitude of a signal increases its dB value by 6 dB. If you kept dividing your sine wave until its peak amplitude was equal to 1, you'd get the very lowest peak dB possible, -90.3 dB. Why do we use dBs? We'll for one, it's easier to say -90 dB than 0.000030 (1/32,768). Decibels have been used for a very long time when dealing with sound pressure levels because of the huge range (about 120 dB) that the human ear can perceive. One confusing thing about using decibels is that 0% is referred to as minus infinity (-Inf. throughout this manual ). How do we measure the levels of a digital signal? Digital meters usually show the maximum instantaneous amplitudes in dB. This is called a peak meter. Peak meters are excellent for making sure that a recorded signal is never clipped. However, peak meters aren't as precise as using RMS (Root mean square…another mathematical formula) power readings when trying to measure loudness. This can be appreciated by generating a sine wave and a square wave with the same peak amplitudes and noting the square wave is much louder. When using RMS power, a maximum-amplitude square wave will be 0 dB (by definition), while a maximum amplitude sine wave reaches only -3 dB. Now, let's get back to the real question – at what level should audio be digitized? If you know what the very loudest section of the audio is in advance, you can set your record levels so that the peak is as close to 0 dB as possible and you'll have maximized the dynamic range of the digital medium. However, in most cases you don't know in advance what the loudest level will be, so you should give yourself at least 3 to 6 dB of headroom for unexpected peaks (more when recording your easily over-excited drummer friend). Now get in there and have some fun with.

About Frames, positions, small frames and bits
The data on an audio CD is divided into frames. A frame consists of 588 stereo samples. 75 frames make up one second of audio. Why? Well 75*588= 44100, and since the sampling frequency of the CD format is 44100kHz (samples per second), this equals one second of audio. When you specify positions on the CD, in Wave-Lab, you do it in the format mm:ss:ff, where mm is minutes, ss is seconds and ff is frames. The frame values go from 0 to 74, since there are 75 frames to a second. Technically, there is no way to specify something smaller than a frame on a CD. One effect of this is that if the length of a Track on the CD does not equal a perfect number of frames, some blank audio must be added at the end. Another effect of this is that when you play the CD, you can never locate (position) to anything closer than a frame. If you need some data in the middle of a frame, you still have to read the whole frame. Again, this is unlike a hard disk, where you can retrieve any byte on the disk, without reading the surrounding data. But frames aren’t the smallest block of data on a CD. There is also something called "small frames". A small frame is a container of 588 bits. 98 small frames to-gether make up one regular frame. In each small frame there is actually only room for six stereo samples, which means that a lot of space is left for other data than the actual audio. There is information for encoding, laser synchronization, error correc-tion and the PQ data (named so "simply" because they are stored in the "P" and "Q" bits). This PQ data is of major importance to anyone who wants to create their own CD, so please let us explain it in further detail.

Another view

A basic primer on sound
Sound travels through the air as longitudinal waves. Molecules of air vibrate, changing their distances between each other. When they come closer, they are known as compressions. When they move further, they are known as rarefactions.
These alternating rarefactions and compressions of air reach your eardrum, and cause the eardrum to vibrate too. Once your eardrum is vibrating, your ear will hear the sound.
Audio signals are measured in Bels, or in the more convenient Decibels (dB), which are one-tenth of a Bel. The Decibel or Bel standard is used to measure sound in the same way humans judge sound. For example, if we were to perceive a doubling of a standard audio volume, the sound level would have increased by about 6dB. However, how much extra energy is required to achieve this doubling would vary according to how loud the sound already was to begin with.
Analogue electric audio signals come in various standards, but typically, they are direct currents that vary in strength. The quickly changing high and low voltages in an audio signal correspond to the rarefactions and compressions of sound, though not necessarily in that order.
When the diaphragms of microphones vibrate, they create a little current into the audio cable. The same can be said for electric guitar pickups, that detect the movement of metal strings over a magnetic field. This current is your audio signal.
A loudspeaker does precisely the same thing in reverse. The current reaches the speaker, and the coils of wire in the speaker turn it back into actual movement, that causes the air surrounding the speaker to vibrate, which results in sound again.
But a microphone produces a very low voltage signal, while a loud speaker relies on a very very high voltage signal to work. The device that sits between the two is an amplifier. The amplifier will increase the voltage in for a higher-power audio signal proportionately to the voltage that it is receiving.

Noise & Distortion
In simple PA Systems, it is possible to work with just one microphone, one amplifier and one speaker. However, for more complicated setups, it might be necessary to alter the sound slightly. For example, karaoke sets add echo to the voice before pumping the sound to the loudspeaker. Also, you may want to add a voice-over to music, just like a DJ speaking over music.
It is overkill to do all this at a signal level powerful enough to drive speakers. To process audio signals that have come straight from the microphone would not be a good idea either, as the signal is barely a trickle of current. It would be like gathering a few scraps of heiroglyphs from an archealogical dig and trying to write a thesis on what the Egyptians thought of Barney the Dinosaur in ancient times. Not only will you have to add a lot of (probably erroneous) information, it will probably be completely different from the truth.
Erroneous sound signals are called noisy, because that's how they will sound. You can often hear this in badly tuned radios as static crackles or a quiet hiss in the background. This arises because your radio is not set up to receive all the audio information to give you a clear sound.
Distortion also damages sound signals. This arises when the equipment producing the sound cannot adequately handle the sounds to give you a fair representation of what it is supposed to sound like. The most well-known example is probably of the Distorted Guitar, the kind of sound produced by heavy metal guitarists. That grungy, grating sound used in guitar solos is interesting, but I'm pretty sure that's not what a guitar actually sounds like. A twang on a guitar string produces a twang, not a Deep Purple power chord.
That previous example also shows that Distortion and Noise are not always undesireable. They can be manipulated to good use.
Clipping is a form of distortion. Clipping occurs when a sound signal's voltage increases past a point that the equipment cannot output. As such, instead of maintaining the real wave-form of a sound, the sound is abruptly truncated where equipment meets its limits. This can sometimes cause the grating-guitar sort of sound.

The Line-Level
To minimise noise and distortion when processing sound, transferring it from one component to another, or reproducing recorded sound, there is a standard type of audio signal called the Line-Level signal. This electronic representation of sound is similar to the types used to drive speakers or coming from microphones and guitars. The only difference is in its strength. It is supposed to be rated at about 1 volt for professional applications, although I have no idea how they rated it. The professional term for this signal is called +4dBu. 
Then some bright spark came along and said, "Hey, why should consumer products use the same sort of signal quality as professional products?" As a result, the totally redundant (but woefully popular) -10dBV signal was invented. This signal is rated at a tenth of a volt. There is actually no reason why there should be two types of signal. The two standards coexist in most studio setups, causing occasional conflicts.
Why was the line-level necessary? It is not so powerful to require high-power equipment to withstand the energy in the signal. In fact, line-level signals come out of just about every home hi-fi system (other than most amplifiers) so that they can be interconnected. It is also not so powerful that it would overload some circuits and cause distortion.
It is also not so low-power that noise begins to obscure all audio detail. As such, line-level signals are most appropriate for transferring audio information from component to component and sending to recording devices. How much power the devices use to record the sound is dependant on the individual method of recording.

The S/N Ratio
One reason why I don't like the -10dBV standard is because the Signal to-Noise Ratio (S/N Ratio) is for the -10dBV standard has to be lower by default. The S/N Ratio is used as a rough gauge to measure how noisy a signal or circuit is. Typically, for a given electronic circuit, the noise in that circuit will remain constant as long as nothing too drastic is done to it. The signal level in the circuit varies according to how much signal level you put into it. Therefore, it stands to reason that if you can get a higher-power clean signal to go into the circuit, there's no reason to use a low-power version, unless the high-power signal is going to overload and distort when it goes into the circuitry. But that is a simple problem that can be fixed with good design. A high-power signal will be less susceptible to noise corruption and cleaner in sound.
The S/N Ratio is basically determined by subtracting the average signal volume from the average noise level. As a result, the S/N Ratio is also measured in dBs. Most consumer equipment have S/N Ratios of 50dB and higher. Anything lower begins to have noticeable noise.
The S/N Ratio will vary depending on the type of signal you are putting into the circuitry. As such, a lot of professional measurements are used by A-weighting. This is measuring the S/N Ratio in comparison to typical audio signals. It does not take into account the S/N Ratio of sounds that only dogs can hear.

Impedance
This is probably the most misunderstood aspect of audio signals. I'm not too sure of my facts myself, but I'll give it a shot. Impedance is a measure of either an output's capability to drive inputs or an input's capability to receive signals from an output. It is measured in Ohms, so it is in some way related to the resistance of audio devices. It is often referred to as Z.
A perfect input device will have no impendance, and a perfect output device will have infinite impendance. In real terms, usually the impedances of outputs are 5 to 10 times that of standard inputs for acceptable performance. Anything less could result in distortion of sound. In effect, the output would not be capable of putting out enough clear signal for the input to pick up properly.
As an example, dynamic microphones typically have impedances of 300ohms to 600ohms. Condenser microphones have impedances in the thousands. Line outputs approach 10kohms. 
Mixer inputs try to have impendances as low as possible to minimize distortion of the sound. Some mixers tout 'Very Low Impedance' or 'VLZ' as a feature of their mixers, which affects microphone inputs more audibly than line inputs. Line inputs are also known as 'Hi-Z' inputs, to accomodate signals that come from high-impedance equipment, i.e. Line-level signals. 
The fact that output impedance is always much greater than input impedance also means that most high-impedance outputs can power multiple inputs at one time. A simple splitter cable would be able to allow two inputs to receive a clear signal from one output, or possibly even more. Thus, the reverse is not true...you shouldn't mix two outputs into one input by using a splitter cable. The impedance would be so badly offset that the sound would be audibly distorted.

Stereo sound
Most humans have two ears, duh. If sound was recorded with two separate microphones and played back with two separate speakers, you could possibly reproduce the same illusion of left-to-right direction for sound reproduction. This is why stereo was created: the common implementation of sounds recorded with two channels: left and right. In order fake the position of a sound somewhere in between, a proportion of the sound is distributed to each speaker.
Stereo sound doesn't succeed in giving the illusion that the band is right in front of you, but it does heighten the aesthetic interest of music or sound recorded. One real benefit of stereo is that you can separate the positions of different instruments, so that each instrument can be heard clearly while being part of a whole mix.

Stereo introduced some new terms of its own: Pan and Balance. Pan, short for panorama, dictates how much of a single sound should be given to the left speaker and how much to the right. This represents which direction in the stereo image the sound should seem to be coming from. Balance stands for how the volume of the left channel compares to that of the right channel in a stereo sound. For example, a stereo recording in which every instrument seems to be closer to the right speaker than the right has not been recorded with the correct balance (unless it was done deliberately). If you were to take a recording of a mono instrument, say, a saxaphone, and put the same sound into the left and right channels of a stereo mixer, adjusting the balance of the left and right channels would actually result in you changing the pan of that single saxaphone. It won't sound like two saxaphones, it would just sound like a single saxaphone moving from left to right as you alter the balance.

Sound Systems

Basics of Sound and Sound Systems 

A MODEL OF A SOUND SYSTEM
Sound systems amplify sound by converting the sound waves (physical, or kinetic, energy) into electrical energy, increasing the power of the electrical energy by electronic means, and then converting the more powerful electrical energy back into sound. Devices that convert energy from one form into another are called transducers. Devices that change one or more aspects of the audio signal are called signal processors. 
The input transducer (such as a microphone or a guitar pickup) converts sound into a fluctuating electrical current that is a precise representation of the sound. This fluctuating current is referred to as an audio signal. 
The signal processing alters one or more characteristics of the audio signal. In the simplest case, it increases the power of the signal (such a signal processor is called an amplifier). In practical sound systems, this block of the diagram represents a multitude of devices-- preamplifiers, mixers, effects units, equalizers, amplifiers, et cetera. 
The output transducer (a speaker or headphones) converts the amplified and processed electrical signal (audio signal) back into sound. 

INPUT TRANSDUCERS

Input transducers, as mentioned before, convert sound into audio signals. Here are some types of input transducers commonly found in sound reinforcement systems: 
Air pressure or velocity Microphones--convert sound waves traveling in air into an audio signal traveling in the microphone cable (see Input Devices for exactly how they do this).
Contact Pickups--convert sound waves in a dense medium (wood, metal, skin) into an audio signal. Sometimes used on acoustic stringed instruments such as guitars, mandolins, violins, etc.
Magnetic Pickups--convert fluctuating waves of induced magnetism into an audio signal. Found on electric stringed instruments (electric guitars, etc).
Tape Heads--convert fluctuating magnetic fields (imprinted on magnetic recording tape (i.e. cassette)) into an audio signal.
Phonograph pickups (cartridges)--convert physical movement of a stylus (needle) into an audio signal.
Laser Pickups--convert imprinted patterns on a compact disc or Mini-Disc into a digital data stream that is then translated by a digital-to-analog converter into an analog signal.
Optical Pickups--convert variations in the density or transparent area of a photographic film into an audio signal. Used for most motion picture sound tracks.

OUTPUT TRANSDUCERS

Output transducers, as mentioned before, convert audio signals back into sound. The following is a list of commonly-found output transducers:
Woofer Loudspeakers--designed specifically to reproduce low frequencies (usually below 500Hz). Woofers sometimes are used to reproduce both low and some mid frequencies. Typically, they are cone-type drivers measuring from eight to eighteen inches in diameter.
Midrange Loudspeakers--designed specifically to reproduce middle frequencies.
Tweeter Loudspeakers--designed to reproduce the highest frequencies.
Full-range Loudspeakers--integrated systems incorporating woofer and tweeter drivers in a single enclosure. As the name implies, they are designed to reproduce the full audio range (more or less).
Subwoofer Loudspeakers--used to extend the low frequency range of full-range systems to include frequencies down to 20 or 30Hz.
Supertweeter Loudspeakers--used to extend the range of full-range systems in the highest frequencies.
Monitor Loudspeakers--full-range loudspeakers that are pointed at the performer on stage, rather than out to the audience. They are used to return a portion of the program to the performer, to help him or her stay in tune and in time, and are usually referred to as "foldback."
Headphones--full-range transducers designed to fit snugly on the ears. Some designs block out ambient (external) sound, while others do not.

The illustration above illustrates a simple, practical sound system that might be used in a lecture hall or media center, etc.
The system can be conceptually analyzed as having three sections: (a) the input transducers, (b) signal processing, and (c) the output transducers:

A] Input Transducers--three microphones convert the sound they pick up from the speakers into audio signals that travel down the cables to the signal processing equipment.
B] Signal Processing-the three microphones are connected to individual inputs on a mixing console. The console serves the following functions:
1] Preamplification-- the console's microphone input section amplifies the level of the audio signal from each microphone, bringing it up to line level. 
2] Equalization-- the console provides the means to adjust the tonal balance of each microphone individually. This allows the console operator to achieve a more pleasing or more intelligible sound quality.
3] Mixing-- the console adds the equalized signals of the microphones together to produce a single line-level output signal. The output of the console is connected to a power amplifier. The power amplifier boosts the console's line level (0.1 to 100 milliwatts) output signal to a level suitable to drive the loudspeaker (0.5 to 500 watts).
C] Output Transducer--the loudspeaker converts the power amplifier output signal back into sound. The level of the sound is much higher than that of the three orators speaking unaided.
There is another less obvious, but equally important aspect of the sound system: the environment. When the sound output of the loudspeaker propagates into the hall, it is altered by the acoustical characteristics of the space. 
The room may have little effect on the clarity of the sound if, for example, the room is "dead" or nonreverberant. If the room is highly reverberant, and the sound system is not designed and installed to deal with the acoustics of the space, the effect on the sound may be so severe as to render the sound system useless. 
The environment is an integral part of the sound system, and its effects must be considered when the system is installed. 
Every sound system, no matter how large, is merely an extension of this basic model. The same principles that apply to this simple model also apply to large-scale concert reinforcement systems. 
Large concert systems may be comprised of twenty stage microphones, twenty keyboard inputs, many drum microphones, maybe a twenty-four track 3324 digital audio tape backup-- but they all follow the same principle: kinetic energy (in the air) is transformed into electrical energy, which is then extensively manipulated and often split to different areas, which may transform the electrical energy back into kinetic energy, or may record the electrical energy.

It should be said here that we believe that anyone working in a technical field... sound, for instance, should have a good background in what sound is, how sound works, what affects sound, etc. In other words, a good background in physics is a good idea. This section will try not to be too technical. But, if it is, check with your local physics teacher to learn more. However, as an addendum after taking a semester and a half of Yale University Physics 200 and a year of Yale University Electrical Engineering, don't overdo it, or at least, if you do, make sure you have no life; i.e. don't do theatre, especially at Yale. 

SOUND

All sounds are created by causing a medium to vibrate-- be it wood, strings, or vocal chords. Sound is carried through mediums by causing adjacent particles to vibrate similarly; the air particles adjacent to a guitar's strings are displaced and "bump" into the next adjacent air particle. This continues and eventually air particles in our ears "bump" into the tiny hairs located in our inner ear. 
The most popular analogy to sound is that of the effect of a rock being dropped into a pond. The ripples, originating from the point source of the rock, spread out in all directions. As with sound, these ripples lose intensity as the distance away from the point source increases. Additionally, these waves form exactly the same shape as a sound wave-- something of sinusoidal curve. 
The distance from a particular point of one wave (be it sound or mechanical) to the same point of the next wave is called the wavelength. Wavelengths of sound range from one inch to forty feet. In a given room, if the distance between two sides of the room is a multiple of the wavelength, this wavelength may be emphasized, which can have either a positive or negative effect. Regardless, we must know how to control it. 
However, in sound, one rarely discusses wavelength. Instead, we count the number of complete cycles these waves can propagate during a specific amount of time, usually one second. This is known as the frequency. Frequency is measured in cycles-per-second, termed "Hertz," abbreviated "Hz". Sounds that vibrate many times per second are known as "high-frequency" sounds, and those which vibrate fewer times per second are known as "low-frequency" sounds. 
The time it takes to complete one cycle is called the period of the wave and is expressed with the symbol T. Thus, T = 1/f. 
Wavelength is usually represented by the Greek letter lambda (which I can't display on WWW html... actually I probably can...) frequency by f, and velocity by v. Velocity is the product of wavelength and frequency, so we get the equation 
v = (lambda) * f. 

PHASE

Since a cycle can begin at any point on a waveform, it is possible to have two wave generators producing the same wave of the same frequency and amplitude which will have different amplitudes at any one point in time. These waves are said to be out of phase with respect to each other. Phase is measured in degrees and a cycle can be divided in to 360 degrees; usually the sine curve is used as an example-- it begins at 0 degrees with 0 amplitude, increases to a positive maximum (the positive amplitude) at 90 degrees, decreases to zero again at 180 degrees, and decreases to a negative maximum (the negative amplitude) at 270 degrees, and returns back to 0 amplitude at 360 degrees. 
Similar waveforms can be added by summing their signed amplitudes at each instant of time. When two waveforms that are completely in phase (0 degrees phase difference) and of the same frequency, shape, and peak amplitude are added, the resulting waveform is of the same frequency, phase, and shape, but has twice the original peak amplitude. If two waves are the same as the ones just described, except that they are completely out of phase (out-of-polarity with respect to each other; phase difference of 180 degrees), they will cancel each other out when added, resulting in a straight line of zero amplitude. If the second wave is only partially out of phase, it would interfere constructively at points where the amplitudes of the two waves have the same sign (both positive or both negative), resulting in a greater amplitude in the combined wave than in the first wave; and it would interfere destructively at points where the signs of the two wave amplitudes are opposing, resulting in a lesser amplitude at those points in time. The waves can be said to be in phase, or correlated, at points where the signs are the same and out-of-phase, or uncorrelated, at points where the signs are opposing. 
Phase shift is a term that describes the amount of lead or lag in one wave with respect to another. It results from a time delay in the transmission of one of the waves. The number of degrees of phase shift introduced by a time delay can be computed by the formula: 
(phase shift) = change-in-t * f * 360 degrees, where change-in-t is the time delay in seconds. 

THE SPEED OF SOUND

Since sound is dependent upon vibration, it can travel through anything except a vacuum. It travels through some materials faster than others; sound travels about four times faster in water than in air, and about ten times slower in rubber. The speed of sound is a very important quantity to know when dealing with large-scale sound reinforcement systems, such as those used in arenas, used outdoors, or over extremely long distances. In air at 0 degreesC and 1 atm (atmosphere- a pressure quantity), sound travels at a speed of 331m/s. Temperature can affect the speed of sound in any medium, but most drastically in gases. In air, the speed increases approximately .60 m/s for each degree Celsius increase: 
v = (331 + 0.60T) m/s., where T=degrees Celsius. 
The speed of sound is virtually constant at all frequencies, but sound will travel faster in humid air rather than in dry air. Humid air also absorbs more high frequencies than low frequencies, so in humid conditions, the sound engineer will need to boost the high frequency portion of the program. 

HOW YOU HEAR

The ear is a nonlinear device and, as result, it produces harmonic distortion when subjected to sound waves above a certain loudness. Harmonic distortion is the production of waveform harmonics that did not exist in the original signal. The ear can cause a loud 1kHz tone to be heard as a combination of tones at 1kHz, 2kHz, 3kHz, and so on. Although the ear may receive the overtone structure (all of the harmonics) of a violin (if the listening level is loud enough), the ear will produce additional harmonics, thus changing the perceived timbre of the instrument. This means that sound monitored at very loud levels may sound quite different when played back at low levels. 
In addition to being nonlinear with respect to amplitude, the ear's frequency response changes with the loudness of the perceived signal. The loudness compensation switch found on many hi-fi preamplifiers is an attempt to compensate for the decrease in the ear's sensitivity to low-frequency sounds at low-levels. The curves below (when I scan them in) are the Fletcher-Munson equal-loudness contours: they indicate the average ear sensitivity to different frequencies at different levels. The horizontal curves indicate the sound pressure levels that are required to produce the same perceived loudness at different frequencies. Thus, to equal the loudness of a 1.5kHz tone at a level of 110 dB SPL, a 40Hz tone has to be 2dB greater in sound pressure level, while a 10kHz tone must be 8dB greater than the 1.5kHz tone to be perceived as loud. Thus, if a piece of music is monitored so that the signals produce a sound pressure level of 110dB, and it sounds well-balanced, it will sound both bass and treble deficient when played at a level of 50dB SPL. 85dBSPL can be considered the optimum monitoring level for mixdowns. 
The loudness of a tone can also affect the pitch that the ear perceives. For example, if the intensity of a 100Hz tone is increased from 40 to 100dB SPL, the ear will perceive a pitch decrease of about 10%. At 500Hz, the pitch changes about 2% for the same increase in sound pressure level. This is one reason that musicians find it hard to tune their instruments while listening through headphones. The headphones are often producing higher SPLs than might be expected. 
As a result of the nonlinearity of the ear, tones can interact with each other rather than being perceived separately. Three types of interaction effects occur: beats, combination tones, and masking.
*Beats: Two tones that differ only slightly in frequency and have approximately the same amplitude will produce beats at the ear equal to the different between the two frequencies. The phenomenon of beats can be used as an aid in tuning instruments because the beats slow down and stop as the two notes are in perfect tune, and the piano tuner will slightly off-tune the instrument by listening to the beat relationships. These beats are the result of the ear's inability to separate closely pitched notes. 
*Combination Tones: Combination tones result when two loud tones differ by more than 50Hz. The ear will produce an additional set of tones that are equal to both the sum and the different of the two original tones and that are also equal to the sum and difference of their harmonics. The formulae for computing the tones are: diff tone frequencies = f1 - f2; sum tone frequencies = f1 + f2, where f1 and f2 are positive integers. The difference tones can be easily heard when they are below the frequency of both the original tones. For example, 2000 and 2500Hz produce a difference tone of 500Hz. 
*Masking: Masking is the phenomenon by which loud signals prevent the ear from hearing softer sounds. The greatest masking effect occurs when the frequency of the sound and the frequency of the masking noise are close to each other. For example, a 4kHz tone will mask a softer 3.5kHz tone, but will have little effect on the audibility of a quiet 1000Hz tone. The masking phenomenon is one of the main reasons that stereo placement and equalization are so important in a mixdown. An instrument that sounds fine by itself can be completely hidden or changed in character by louder instruments with a similar timbre. 
Although one ear is not able to discern the direction from which a sound originates, two ears can. This ability of two ears to localize a sound source within an acoustic space is called binaural localization. This effects results from using three cues that are received by the ears: interaural intensity differences, interaural arrive-time differences, and the effects of the pinnae (outer ears). 
Middle- to higher-frequency sounds originating from the right side will reach the right ear at a higher intensity level than the left ear, causing an interaural intensity difference. This occurs because the head casts an acoustic block or shadow, allowing only reflected sound from surrounding surfaces to reach the left ear. Since the reflected sound travels farther and loses energy at each reflection, the intensity of sound perceived by the left ear is reduced, with the resulting signal being perceived as originating from the right. 
This effect is relatively insignificant at lower frequencies, where wave-lengths are large compared to the diameter of the head and easily bend around its acoustic shadow. A different method of localization known as interaural arrive-time differences is employed at lower frequencies. In our example, time differences occur because the acoustic path length to the left ear is slightly longer than that to the right ear. The sound pressure will thus be sensed by the left ear at a later time than by the right ear. This method of localization, in combination with interaural intensity differences, gives us lateral location cues over the entire frequency spectrum. 
The intensity and delay cues allow us to perceive the angle from which a sound originates, but not whether the sound originates from the front, behind, or below. The pinna, however, makes use of two ridges that reflect the incident sound into the ear. These ridges introduce time delays between the direct sound (which reaches the entrance of the ear canal) and the sound reflected from the ridges (which varies according to source location). 

PITCH

The pitch of a sound refers to whether it is high, like the sound of piccolo or violin, or low, like the sound of a bass drum or string bass. The physical quantity that determines pitch is the frequency. The lower the frequency, the lower the pitch. The human ear responds to frequencies in the range from about 20Hz to about 20,000Hz. This is called the audible range. These limits vary somewhat from one individual to another. One general trend is that as people age, they are less able to hear the high frequencies, so that the high-frequency limit may be 10,000Hz or less. 
Sound waves whose frequencies are outside the audible range may reach the ear, but we are not generally aware of them. Frequencies above 20,000Hz are called ultrasonic. Many animals can hear ultrasonic frequencies; dogs, for example, can hear sounds as high as 50,000Hz and bats can detect frequencies as high as 100,000Hz. 
Sound waves whose frequencies fall below the audible range are called infrasonic, or occasionally subsonic. Sources of infrasonic waves are earthquakes, thunder, volcanoes, and waves produced by vibrating heavy machinery. 
The pitch of the sound also factors into the way the ear hears. The ear has difficulty in associating a point origin to a low-frequency sound, but is quite accurate in placing the origin of high-frequencies. This is because high frequencies have wavelengths shorter than the distance between the ears; sounds above 1000Hz cannot reach both ears at the same time and at the same intensity, so one ear is favored and provides the information as to the direction in the horizontal plane. The ear is less successful in responding to directions in the vertical plane. 

FUNDAMENTALS AND HARMONICS

The initial vibration of a sound sources is called the fundamental, and thus the initial frequency is known as the fundamental frequency. The subsequent vibrations, which are exact multiples of the fundamental frequency, are called the harmonics. So, a note on a musical instrument with a fundamental frequency of 100Hz will have a second harmonic at 200Hz, a third harmonic at 400Hz, et al. 
The term octave denotes the difference between any two frequencies where the ratio between them is 2:1. Thus, an octave separates the fundamental from the second harmonic in the above example: 200Hz:100Hz. At the upper end of the frequency spectrum the same ratio still applies although the frequencies are greater. An octave still separates 2000Hz from 1000Hz. Two notes separated by an octave are said to be "in tune." Thus, an octave on the piano keyboard, separated by eight keys (well, really thirteen), is also an octave-- frequency-wise. 
Whether the harmonics diminish in intensity or retain much of their energy depends on how the source is initially vibrated and subsequently damped. It is the strength of the harmonics which distinguishes the quality (or timbre) of musical instruments and makes it possible for humans to identify two different instruments playing the same note. Cool, huh. 

INTENSITY

Like pitch, loudness is a sensation in the consciousness of a human being. It, too, is related to a physically measurable quantity, the intensity of the wave. Intensity is defined as the energy transported by a wave per unit time across unit area. Since energy per unit time is power, intensity has units of power per unit area, or watts/meter2 (W/m2). The intensity depends on the amplitude of the wave (it is proportional to the square of the amplitude). [The amplitude of the wave is the distance between the extremes of the vibration.] 
The human ear can detect sounds with an intensity as low as 10-12 W/m2 and as high as 1 W/m2 (and even higher, although above this it is painful). This is an incredibly wide range of intensity, spanning a factor of 1012 from lowest to highest. Presumably because of this wide range, what we perceive as loudness is not directly proportional to the intensity. Ture, the greater the intensity, the louder the sound. But to produce a sound that sounds about twice as loud requires a sound wave that has about ten times the intensity. For example, a sound wave of intensity 10-9 W/m2 sounds to an average human being as if it is about twice as loud as one whose intensity is 10-10 W/m2; and an intensity of 10-2 W/m2 sounds about twice as loud as 10-3 W/m2 and four times as loud as 10-4 W/m2. 
Because of this relationship between the subjective sensation of loudness and the physically measurable quantity intensity, it is usual to specify sound intensity using a logarithmic scale. The unit on this scale is the decibel, (dB). The intensity level, b, of any sound is defined in terms of its intensity, p, as follows: 
b(dB) = 10 log (p1/p0).
p0 is the intensity of some reference level. It is usually taken as the minimum intensity audible to an average person, the "threshold of hearing," which is 1.0x10-12 W/m2. Notice that the intensity level at the threshold of hearing is 0dB; that is, b=10 log (10-12/10-12) = 10 log 1 = 0. Notice, too, that an increase of intensity by a factor of ten corresponds to a level increase of 20dB. Common loudness levels and intensity levels for common sounds follow. 
It may be noted that the previous description of decibels is simply a brief overview, something that one would find in a generic physics text. A more in-depth look at decibels is needed for use in sound reinforcement. 

dB IN GENERAL

The dB always describes a ratio of two quantities. Remember that. It's not really important for you to grasp the logarithm concept just now (but if you do, that's cool)... it's simply important that you realize that a logarithm describes the ratio of two powers, not the power value themselves. To demonstrate this, let's plug in some real values in the dB equation. 
dB SPL
This is the one of the more common forms of the decibel. It measures sound pressure levels (SPL): the sound pressure is the level measured per unit area at a particular location relative to the sound source. When a dB describes a sound pressure level ratio, a "20 log" equation is used: 
dBSPL = 20 log (p1/p0), 
where p0 and p1 are the sound pressures, measured in dynes per square centimeter or Newtons per square meter. 
This equation tells us that if one SPL is twice another, it is 6dB greater; if it is ten times another, it is 20dB greater, and so forth. 
How do we perceive SPL? It turns out that a sound which is 3dB higher in level than another is barely perceived to be louder; a sound which is 10dB higher in level is perceived to be about twice as loud. Loudness, by the way, is a subjective quantity, and is also greatly influenced by frequency and absolute sound level. 
SPL has an absolute reference value (p0); generally 0db SPL is defined as the threshold of hearing in the ear's most sensitive range, between 1kHz and 4kHz. It represents a pressure level of 0.0002 dynes/cm2, which is the same as 0.000002 Newtons/m2. It is really best to compare SPLs with each other, as in the following chart.

dBW

We have explained that the dBm is a measure of electrical power, a ratio referenced to one milliwatt. dBm is handy when dealing with the miniscule power (in the millionths of a watt) output of microphones, and the modest levels in signal processors (in the milliwatts). One magazine wished to express larger power numbers without larger dB values... for example, the multi-hundred watt output of large power amplifiers. For this reason, that magazine established another dB power reference: dBW: 
dBW = 10 log (p1/p0), 
0 dBW is one watt. Therefore a 100 watt power amplifier is a 20 dBW amplifier (10 log (100/1) = 10 log (100) = 10*2 = 20dB.) A 1000 watt amplifier is a 30 dBW amplifier, and so forth. 
dB PWL
Acoustic power is expressed in acoustic watts, and can be described with a dB term, dB PWL. This term shares the same "10 log" equation as other power ratios: 
dBPWL = 10 log (p1/p0), 
Acoustic power and dB PWL come into play when calculating the reverb time of an enclosed space, or the efficiency of a loudspeaker system, but they are seldom seen on specification sheets and seldom used by the average sound system operator. It is much more common to use dB SPL because the sound pressure is more directly related to perceived loudness (and is easily measured). 
Incidentally, there is no set relationship between dB PWL and dBW; the former expresses acoustic power, the latter electrical power. If a loudspeaker is fed 20 dBW, it might generate as little as 10 dB PWL. In English... feed 100 watts into a loudspeaker, it might generate as little as 10 watts of acoustic power. This would indicate a conversion efficiency of ten percent, which is high for a cone loudspeaker in a vented box! 

RMS

"RMS" is an abbreviation for a term known as "Root Mean Square." This is a mathmetical expression used in audio to describe the level of a signal. RMS is particularly useful in describing the enegry of a complex waveform or a sine wave. It is not the peak level, nor the average, but rather it is obtained by squaring all the instantaneous voltages along a waveform, averaging the squared values, and taking a square root of the number. 
The rms value of a periodic function, such as the sine curve, is .707 times the peak value of the wave. 
Why is the rms value of a signal used? For one thing, the rms value correlates well with the real work being done b y the amplifier. When so-called "program" or "music" power ratings are employed, the actual work being done is subjective-- it depends largely on the nature of the program source. The rms value of any program will pretty much reflect the energy content of that source. There is just one minor problem: the term "rms power" is meaningless. 
Why? Power is the product of voltage multiplied by current. Typically, in a power amp, one is measuring the rms value of the output voltage and multiplying it by the rms value of the output current. This does not result in the rms power because the voltage and current are not in phase, and hence the rms values do not multiply to form a mathematically valid value. The intent of an rms power rating is valid, but not the term itself. Manufacturers are still driving amplifiers with sine wave test signals and connecting the amp outputs to dummy loads. They obtain the rms value of the sine wave output based on that voltage and the load resistance or impedance. Those who wish to be technically correct list this rating as "continuous average sine wave power," rather than "rms power." 
RMS values are not the exclusive domain of power amplifiers. In most (but not all) cases, when you see a voltage listed for input sensitivity on a preamplifier or line amp, it is the rms voltage. For example, you may recall that 0 dBm is 1 milliwatt, which equals 0.775 volts rms across a 600 ohm circuit, and 0 dBV is 1 volt rms. 

WIRES AND CONNECTORS
Wires and connectors link audio components together to form an audio system. Although there are quite a number of different standard connectors and cables, they all carry more or less a variation of the same type of audio signal: a voltage fluctuation corresponding to sound waveforms. The only major differences arise when comparing analogue sound to digital sound.

Shielding
Audio voltages are sometimes measured in millivolts and fractions of a volt. At such low voltages, it is highly susceptible to interference from outside sources. Such interference is sometimes created by the existence of electromagnetic radiation around the audio cable. EM radiation can result from any nearby electrical components, cables, or even neighbouring radio stations. Components that utilize large amounts of energy, fluorescent and halogen lights and those that have motors or transformers are usually the worst culprits. The current in the circuits of these components create a fluctuation electromagnetic field around the component. This field in turn induces undesirable currents in neighbouring circuits. If the neighbouring circuits are audio cables, the result can sometimes be audible as hum, noise or occasionally distortion.
In order to avoid interference, insulated audio cables are usually surrounded by a layer of shielding. This is a flexible metal cylinder of wire or foil that surrounds the actual audio cable. The whole bundle of shielding and insulated audio cable is usually surrounded by another layer of insulation. 
The theory is that the metal layer formed by the shielding will absorb most of the influence of EM fields, reducing the impact of EM radiation on the signal cable inside. The resulting audio signal would be freer from the noise and hum of EM fields. This theory works reasonably well for areas of low EM radiation.
Some cables are arranged such that there is one signal conductor in the centre surrounded by a cylinder of shielding. These are sometimes termed co-axial cables, because the cross-section of both wires have the same circular centre (axis). This means that the signal cable is about the same distance away from the shielding at any point in the cable. This is desirable because that means that the shielding capability of the cable is uniform across the whole cable. It also means that if the shielding does carry a current of its own, the EM fields generated by the current should automatically cancel out when it reaches the signal cable, and vice-versa.
Cables with one signal conductor and one shielding layer are called unbalanced. The exception is with stereo cables, in which two signal carrying cables may share one shielding. Although they have two signal conductors, they are also carrying two separate signals, so they are in effect two unbalanced cables in one.
Shielding and weight
Shielding is usually found as either a foil sheath under the top layer of insulation of a cable, a twisted coil of wire around the inner cable, or certain expensive cables use a conductive plastic/graphite insulator as shielding. Professional mic cables usually employ a braided sheath of copper wire, as this is not easily unravelled and lends strength and rigidity to the cable. Foil is a good shielding for cables that do not see much rough use, as it is thin and light. It is especially appropriate in a studio situation, where a patchbay can have 40 or more cables running into it. If braided heavy mic cable was used, the shielding would probably be very effective but the weight of over 40 cables would probably pull the patchbay off the rack.
Balancing
Shielding has its limitations. Strong EM fields can still affect the inner signal cable to an audible degree. As such, a second method of preventing interference, known as balancing, is used in professional applications and low-voltage equipment (microphones, for example). This uses two cables to carry one signal and usually one layer of shield.
One cable is known as 'Hot', the second is known as 'Cold', and the shielding is known as 'Shield' or 'Ground'. The 'Hot' and 'Cold' is usually used for unbalanced signals, in which one cable carries the signal and the other serves as a return path of the signal. However, since the convention stands, the naming is the same for the balanced cables. In balanced cables, both serve as signal carries for inverted signals.
The shielding in balanced cables acts as normal, but the two insulated cables are twisted together and kept as close together as possible. The theory is that, as the two cables are kept as close as possible to each other while still being insulated from each other, any EM radiation that penetrates the shield will affect both cables equally.
In order to remove the artifacts produced by the interference, one of the signals can be reversed, and mixed with the other signal. According to high-school physics, if the artifacts on both cables are identical, the reversal and mixing of the signals will cancel out the noise.
Of course, if the signals between the two signal cables are also identical, the reversal and mixing of the signals would also result in a cancellation of the signal, which means you won't get any sound at all. Duh. Therefore, before the signal is introduced into the cable, one is already reversed. When the reversal occurs again at the receiving end of the cable, the signal strength would be doubled, and the noise should be cancelled. 
This method of noise cancellation works rather well, especially if the source of the EM fields are some distance away from the cables. If the field source is too close, or if the area surrounding the cable has varying EM absorption and reflection properties, then the EM field imposed on the two signal cables may not be identical, and the noise cancellation would be less effective.

Ground Loops
According to electrical physics, a loop of wire placed in an area with a fluctuating EM field could quite possibly have a current induced in it. In audio terms, your cables could quite possibly be generating current due to the EM fields in the vicinity. A loop in the cables would allow the noise or hum to increase in power quite considerably, resulting in noise or an audible. But in most installations, signals are not looped. Doing so would result in feedback, the high pitched wail that you hear when you stick a microphone near its own speaker.
Grounding and shielding in the cables, though, are another matter. Equipment that require large amounts of power will probably be grounded, especially if your mains voltages are 220V instead of the American 110V standard. This will reduce the possibility of a fatal shock should your equipment short circuit for any reason. Everything is connected to a single mains earth, which is usually connected to all the earth pins in all the power sockets in one room.
This would normally be okay, as the grounding is only connected to each other in a star-like fashion. From a central earth wire (leading to the real Earth via a grounding cable or metal pipe) earth cables run through your power cables into the equipment. Once you take into account that some of your equipment is linked with shielded cables, the equation gets more difficult.
Currents could quite possibly run from one piece of equipment, into the earth cable, into another piece of equipment, then back to the first piece via a shielded audio cable. The result is similar to feedback: the unwanted signal from the current will be amplified until it is audible and clearly undesireable.
Getting rid of ground loops are not easy. Finding them is even harder. The only acceptable method is to remove and disconnect everything, and reinstall the equipment piece by piece. Stop when the hum becomes audible, and fix the problem in the last piece of equipment that was installed. It is possible to solve the problem by systematically unplugging and replugging cables until the change in one particular cable results in a dramatic change in hum, and fix it there, but this is purely a short-term solution. Any further change to the set-up may result in the loop reappearing.
To kill a ground loop, the common-sense method would be to break the loop. Removing the earth cables from the equipment to the mains is not recommended, but it works. You just have to be careful not to electrocute yourself.
Breaking the shielding of a cable is possible, but it renders the cable susceptible to the influences of EM fields. However, breaking a cable is effectively introducing an infinite resistance into the shield. How about introducing just a little resistance into the shield?This actually works to some degree. Some connectors might be able to accomodate a small resistor into the casing. The shield of the cable would thus be soldered to the resistor, and the resistor soldered into the shield contact on the connector. A resistor of about 100ohms would be sufficient. What it actually does is reduce the current produced by a ground loop to a level where it is hopefully inaudible, even though it still exists.
If the effects of EM fields become audible, it might also work to introduce a 4 pF capacitor parallel to the resistor (where you have space for this, I have no idea). This would even out the currents so that instead of fluctuating and becoming audible, they would remain at a similar level continuously.
Actually, a well designed piece of equipment will have a ground lift. This is a 100ohm or so resistor between the mains ground of the equipment and all the shielding contacts of a piece of equipment. This would reduce possibilities of audible ground loops. A truly covers-all bases piece of equipment would have a switch to activate or deactivate the ground lift. In real life, some DI boxes have this facility, in their quest to match levels and reduce interference. To test if your equipment has ground lift, insert a balanced TRS or XLR cable into the equipment, and measure the resistance between the shield contact of the exposed connector and the casing of the equipment or the earth pin of the mains cable of the equipment, using a multimeter. If there's a 100ohm to 500ohm resistance, your equipment is ground lifted.

Speaker Cables
Speakers require far more power than microphone or line-level inputs. As such, they require cables with a thicker guage of metal wire, if not, they might overheat and burn up. Also, since the signal power is so large, noise is much less a problem, so shielding is not so common on speaker cables, although shielded speaker cables do exist (I know, I use them). Speaker cables come in all shapes and thicknesses, so it's hard to typify them. Generally, cables for microphones or guitars are thick and smooth, due to the shielding and the insulation. Cables of line signals can sometimes be thinner, due to lack of balancing or foil shielding, although balanced line signals can be carried in cables as thick as mic cables. Speaker cables are either flattish (for home installations, to avoid a messy looking cable), bumpy (due to lack of insulation and a thick wire guage) or even just a pair of black and red wires twisted together (the most flexible and easily replaceable, always potentially a tangle hazard). 

Types of Cable
Cables used to transfer audio information are usually made of metal. Cheap cables could be the type of copper cables you might expect to find in lighting or mains wiring. Usually cheap cable could be pressed into service in a hurry, especially when using bare-wire connectors. This of course also implies that the cheapest cables are used for speaker connections. As above, ensure that speaker cables are capable of handling the high electrical power of speakers or they might melt.
More expensive types of metal cables, in order of cost, range from Oxygen-Free Copper (OFC), Silver and Gold. OFC is longer lasting than plain copper and allows less distortion to corrupt the audio signal passing through it. Silver cables and Gold cables are usually reserved for audiophile applications, and the improvement in sound is usually minimal for the casual listener.
There are non-metal cables for audio, typically not used for speaker connections. Graphite, the stuff that makes pencil 'lead' black and writeable, can actually conduct an electric current when it is passed in a certain orientation relative to its atomic structure. Graphite cables are available, but are very difficult, if not impossible to solder on your own. They have to be made at the factory. They are supposed to be less susceptible to incidental electromagnetic fields than metal cables, and do not bias the sound in any way. In effect, a very neutral, distortionless cable. Also very expensive.
Plastic polymer cables are also available, but I have not come across them yet. I'll update this page with relevant information when I get the chance.

Connectors
There are a number of different types of connectors used to affix cables to equipment. Balanced equipment requires at least three separate points of contact from the cable to the equipment, while unbalanced equipment requires at least two. Stereo unbalanced connectors requires at least three. Confused? Read on.

Jack or connector?
What is the difference between a jack or a connector? Although terminology varies from place to place, a jack is usually used to refer the outlet or inlet on equipment that allows a cable to be inserted and connector. A connector is found on the ends of a cable. A cable connector will plug into an appropriate equipment jack. Both jacks and cables can come in female and male versions.

XLR connectors
XLR connectors come in several names: Switchcraft, Cannon, and XLR. They are all basically the same. The difference is in the manufacturer of the connector and how the connector is built. All different forms of XLR can be readily connected with another XLR jack or connector.
XLRs have an interesting male-female convention. Whereas most other connectors just have female on the jacks and male on the connectors, XLR can have either on either side. Although this seems like an unnecessary hassle, it is actually an advantage of XLR. The rule of thumb is: if the jack/connector is to be connected to a source of audio signal, then it should be a female. If the jack/connector is to be connected to a receiver of an audio signal, then it is a male. This is an easily understood convention for anybody past puberty, so equipment with XLR connectors are usually wired up correctly (with occasional blunders). Cables typically would have male on one end and female on the other, thus making the extension of XLR cables really easy: just plug in another cable. No adaptors required.
XLR is also a very secure type of connector. A well-made XLR connector would have a metal cylinder with three pins inside. The pins are arranged in an isosceles triangle to avoid connecting the cable the wrong way round. The pins connect up the shield, the male and the female conductors, and are usually clearly numbered on the connector. The cylinder is not an electrical signal carrier at all, it is just a rigid protective covering. A clip at the top of the female cylinder fits into a rectangular gap on the male cylinder, making a secure and audible 'click' when the two parts are connected. The connectors/jacks cannot be removed once a connection is made unless the clip is depressed, or a very strong force is exerted on the cable, in which case, the cable will probably snap at the same time. A rubber or plastic tube forms a grip for the cable so that it is not easily broken off the connector.
As you might have deduced by the three pins, XLR is typically a connector for balanced cables. An XLR-to-unbalanced cable can be made by connecting the shield with the cold conductor at any point of the cable, but this would lose the signal strength and the noise cancellation abilities of the balanced connectors. 
Unfortunately, there is a strange variation of the XLR standard that exists in some European standards. This mixes up the connectors such that the first pin is the hot conductor, the second pin is the cold conductor and the third pin is the shield.
Although there are pre-made XLR cables on the market, XLR jacks are usually sold separately for professionals to solder to cables of their choice of length and type. Cannon, Switchcraft and Neutrik are three well-known brands of XLR connectors. Cannons are the originals, but I haven't had much experience assembling these.
Switchcraft connectors are more durable, but when disassembled, separate into several small pieces. This could make assembly a little trickier. For example, the spring for the clip-button on the female connectors can fall out of the assembly when the connector is unscrewed. Lose anything and you'll have an imperfect connector. The durability of Switchcraft connectors is far superior to most other connectors. Miniature screws, the type only accessible to jeweller's screwdrivers, are used extensively in the assemblage.
Neutrik (of Switzerland) make the easiest to assemble XLR connectors. They come in about 4 easy-to-handle pieces. Grooves inside the parts help each piece slide in the correct direction and orientation. No screwdrivers are required for assembly. The metal casing seems a little thinner than Switchcraft's, but don't dent easily unless (like I have) thrown from two or more stories onto concrete. Note that a dented male connector is typically a useless connector, since it's almost impossible to un-dent it and the female connector will never fit with a damaged male connector.
Neutrik's connectors have cable grips that, although easy to assemble (just screw them in), don't seem as rock-solid as Switchcraft grips. Neutrik connectors may be a little less resistant to tension in cables, but I have had no problems with them in the past.

1/4" or Phone Connectors
These are the big-brothers of the miniature stereo plugs that are found on Walkmen and smaller mini-compo systems. They are found in one long shaft 1/4" in diameter. Along the shaft there are divisions of plastic that separate the contact points along the shaft. There is also a little constriction in the shaft that permits a 1/4" connector to lock into a jack when inserted, though this is nowhere as secure as an XLR locking clip. Since they are usually used for headphones, they are also called Phone connectors.
1/4" Connectors come in two-contact and three-contact connectors. I have seen a four-contact connector somewhere before, but I can't remember where. Two-contact connectors are strictly mono-unbalanced connectors. Three-contact connectors, often found on larger headphones, could either be used for stereo-unbalanced connections or mono-balanced connections.
Two-contact connectors are commonly termed as TS connectors, as the shaft is divided into two parts: the Tip, and the rest of the connector, known as the Sleeve. The constriction in the shaft is part of the tip in both two- and three-contact connectors. The tip is usually used for the signal ('hot') and the sleeve is used for shielding and the signal return ('cold' as well as 'ground').
Three-contact connectors are known as TRS connectors. The shaft has an additional plastic division in would have been the sleeve in a two contact connector. The contact closer to the tip is known as the Ring, while the contact closer to the cable is known as the Sleeve. The Tip is the 'Hot' connection, the Ring is the 'Cold' connection, and the Sleeve is the 'Ground' connection in balanced TRS connectors. For stereo connectors, the Tip and Ring are both the 'Hot' contacts for the left and right signals respectively, while the Sleeve is both the 'Cold' and 'Ground' for both sides.
1/4" connectors are usually male, and jacks are usually female. Interconnection between two cables usually require a female-female adaptor (a little cylinder to whole two shafts).
A variation of the stereo 1/4" connector is found in the Y-splitter. This consists of two unbalanced cables connected to one TRS 1/4" connector. Instead of carrying left and right signals, the Y-splitter carries one signal into a jack and another signal out. This is used in mixer inserts, in which a signal is tapped out of a mixer for processing with equalisers, echo units, and other processors, then put back into the mixer to continue the signal path. With a Y-splitter, the 'pulling out' and 'putting back' of a signal can be achieved with one TRS jack.
Some 1/4" jacks have a special feature that, as far as I know, can only be found on mini-phone jacks as well. This type of jack is known as the break-jack. One example of its use can be seen in some stereo systems, in which the insertion of a headphone (1/4" connector) results in the cutting-off of the main loudspeakers. This jack has a slightly more complex design, but basically it permits a signal to travel in one set of paths, or you can insert a 1/4" connector and have it travel through the connector while cutting off the original set of paths. Thus, a connector inserted into a break-jack becomes both a switch and signal connector at the same time.
It is possible to 'tap' or 'sniff' a signal out of a break-jack without cutting off the original flow of signal by partially inserting the connector into a break-jack, such that it doesn't 'click' into place. Of course, this means that a little jarring will cause the 1/4" jack to fall out of place, so this practice isn't really recommended.
There's a whole range of different manufacturers for TS and TRS 1/4" connectors out there. I prefer the cable brand Canare, which is mostly metal. I prefer these because, if you slip with the soldering iron, you don't inadvertently melt any plastic parts. There is also a good spring type cable support which prevents cables from bending too sharply as they come out of the connector, and a supplied little plastic tube with each connector that prevents the soldering joints from coming into contact with the shield. They are a little on the large side, which might be a problem in some cases, but have generous room inside the connector for soldering.
Some people prefer plastic-clad connectors, since you can't get electrocuted from just holding the connector (very very rare...it means your shield is carrying a major current), the shields of adjacent connectors cannot accidentally touch (leading to ground loops) and they are cheaper and lighter. I've found them to be less durable and difficult to re-solder if you want to reuse pieces, but use what you prefer.

RCA/Phono Connectors
These are the most common connectors in home hi-fis, CD players, LD players and Videocassette recorders. They are simple in construction and are strictly unbalanced. RCA jacks are a small cylindrical stub with a hole in the middle. RCA connectors have a small pin about 2-3 mm in diameter with a cylindrical 'flower' of metal around it. Connectors are typically male and jacks are typically female.
RCA jacks and connectors have an inner contact (pin-to-hole) and outer contact (flower-to-cylinder). The inner contact is for the 'hot' signal and the outer contact is for both the 'cold' and 'shield'.
RCA jacks are typically used for carrying -10dB line-signals, although they are also employed for phonographs/turntables, which is why RCA jacks are also known as Phono jacks. Don't get these mixed up with Phone jacks! They could also quite possibly carry +4dB signals, although this is rarer.
RCA connectors are known to be a little less reliable than 1/4" and XLRs, especially the plastic pre-made type. They are not designed for heavy-duty use, although some more expensive RCA connectors can last quite a long time. For example, Monster Cable builds a special type of RCA connector that has a slitted turbine-like flower that is ridiculously difficult to remove and insert (which was the point in the first place) while Canare's all-metal RCA connectors are almost as long as the 1/4" connectors they make, putting a little more stress on the jack than the connector. Although RCA connectors are easily dented, they are also easily bent back, although I wouldn't recommend using a damaged connector for a long period of time, as it would probably fail again unexpectedly.
I personally find 1/4" connectors easier to solder than RCA connectors, as the RCA connector would typically use a mix of the 'needlehole' solder joints (common in 1/4" jacks) and the 'spade' type of solder joints (common in XLRs). I like 'needlehole' joints as long as your cables are clean and well trimmed, otherwise they would just be too fat to fit in the needleholes for soldering.
RCA connectors range from the super-cheapo flimsy plastic whatsits to the mega-expensive overkill metal and space-age composite things in audiophile shops. Buy according to your needs, don't go overboard: the difference in sound quality will probably be nearly inaudible. For almost-permanent setups, the cheap, pre-made cables will be sufficient. For setups that require constant changing and rearrangement, you may want to spend a bit more on something that doesn't weaken or rust easily.
RCA connectors are also used in digital audio as the jacks for digital coaxial cables. These are usually slightly higher quality with gold plated terminals. Composite video can also be carried over RCA connectors, so most VCRs and TV sets should be able to support both sound and video being carried over separate RCA cables. These cables are usually color-coded with yellow being the video connector, white or black being the stereo-left sound connector, and red as the stereo-right audio connector. In a pinch, a stereo pair of RCA cables can be rushed into service for video and mono-audio, with little or no difference in either audio or video quality.

Mini-phone connectors
Named because they are used most commonly for miniature earphones, these smaller versions of the 1/4" connectors require the least space of all the popular connectors. They are ubiquitous on Walkmen, Discmen, Watchmen, Camcorders, PC sound cards, Macintoshes and other portable audio or video components. Really small music sequencers, such as the Yamaha QY10, also utilise these connectors.
These connectors are really unreliable. The springs in the jack have been known to give up after a dozen insertions and removals on budget equipment. They should only be used on really small equipment, and if possible, on connections that need not be removed and re-inserted often. 
It is thought that these components are harder to solder than RCAs, but I've found them about the same difficulty. Internally, the joints in a mini-phone connector are not much smaller than RCA solder joints. Like their big brothers, they also come in plastic and metal versions. The plastic versions tend to melt a little when soldering and are practically unuseable if you try resoldering new cables to it twice. 
They come in TRS connectors and TS connectors. The joints are exactly the same in positioning as the 1/4" version, and break-jacks are also possible for mini-phone jacks.
Mini-phone jacks are also used for Plaintalk microphones on Macintoshes, but these are slightly longer in length. Although they are TRS jacks, the contact points serve slightly different purposes. Most importantly, the tip of the Plaintalk connector is used to draw power out of the Mac to power electret Plaintalk microphones. Dynamic microphones and line-level TRS connectors can also be used in a Mac, as they are not long enough to reach the contact point that supplies power. Condensor microphones have to draw their power from somewhere else, maybe a battery pack.

Other connectors
That's it for the whole list of common connectors for analogue audio signals. There are a few others that aren't as common, but have been seen used in enough in equipment to warrant a mention.

Banana Plugs
These connectors are just a more flexible alternative to bare wires. Typically, they only have one big conductor, which is a pin with several flat springs on the side. A banana jack is just a hole, in which the banana connector slips into and fits snugly. It is less prone to fraying than bare wires and simple to use for quick installations and deinstallations. It can usually withstand light tugs, but a hard jerk would definitely pull out the connector from the jack. Certain bare wire jacks (the screw-tight type) also have a hole in the middle of the screw to accomodate banana plugs. As far as I know, they are only used for speaker cables.

Leads/Bare Wires
The simplest type of connectors, usually only seen on speaker cables. Bare wires can be connected either by screwing down a jack so that the cable is held tight, or by inserting the wires while holding down a spring-loaded catch which would grip the wires tight when released. Bare wires have a tendency to fray and break after repeated use. Soldered wire leads don't fray as easily, but screw-tight connections might lose their tightness after time, as solder still flows even when cold, at a much slower speed, of course, than liquid solder.
Leads are shaped pieces of metal soldered to the ends of cables. They are just designed to overcome the fraying problems of cables. Two types of leads are the 'spade' and the 'clamp'. The 'spade' looks like a blunt two-toothed fork, designed to fit in screw-tight jacks. 'Clamps' are common on batteries. The 'clamp' jack is just a flat piece of metal. The 'clamp' connector is a metal piece in the shape of a horizontal 'B' meant to grip the jack tightly.

Speakon Connectors
This is a new, heavy duty type of connector designed specially for speaker cables, because of the heavy power loading of such applications. They are highly secure and made of plastic, in order to prevent shock hazards from touching the connectors. To secure a Speakon connection, you must insert the Speakon connector to a Speakon jack, twist the Speakon connector about 45°, then tighten a ring around the connector that forces the connector towards the jack. As such, the cable would probably be severely damaged by a strong cable jerk before the Speakon even begins to budge.
I've tried soldering them, but it's almost impossible, as the contact points are surrounded by plastic and melting the plastic is unavoidable. To utilise them, there are hex screws on the side in which you tighten onto bare cables. The hex screws are tiny, so you might need to get a special Allen key to tighten them. I usually use bare wires, without tinned/soldered tips, as I don't want the tightening to weaken after time. They are designed to last a long time.
 

Digital Signals
Digital signals are communicated from machine to machine typically as a series of pulses of 'on' or 'off' signals, high voltage or low voltage signals, instead of a continuous fluctuation like analogue signals. Digital cables are usually short. When they are metal, they could be high-quality OFC, silver or even gold cable. Graphite cables are also sometimes used for digital signals.
Digital signals are usually electrical, and they use much the same type of connectors as analogue signals. Again, the connectors are of higher quality, for instance, they might be gold-plated. RCA, mini-phone and XLR connectors are frequently used for digital connections.
However, digital signals might be optical. A small laser Light Emitting Diode at the output end of a digital audio component and a small Light Sensitive Resistor at the input end allow blinking lights to be transferred down optical fibers. Of course, light is hardly affected by the electromagnetic radiation of a typical room, so corruption of the signal is even less of a possibility. The connectors used for fiber optics would have one transparent end, that fit inside special jacks in the audio components.
The question is, with the supposed higher resistance to corruption that digital signals are supposed to have, why do they require such higher quality components? Perhaps it is due to the fact that unlike analogue signals, a partial loss of digital information would effectively render the whole signal being unuseable. A partial loss of analogue signal would either result in a lower S/N ratio, or a loss in the high frequency part of the sound, but the sound might still be useable.

Understanding amplifier power ratings
There are different methods for measuring the power ratings for amplifiers and speakers. And different measuring methids give different values so it is vital to understand the difference between theosedifferent power ratings to be able to make at least some comaparisionf between different power ratings. This article is collection of information posted to rec.audio.tech newsgroup at July 1996. The information is compiled from Usenet newsgroup rec.audio.pro articles written by Norbert Hahn, Dick Pierce and Earl K. 
 

RMS power
To make it short, an RMS power value is directly related to perceivable energy (acoustical, heat, light - or what else applies). 

"RMS" is really a rather meaningless figure, when measuring power. R.M.S. is useful for measuring the "power-producing equivalent" voltage. Thus 10 Volts RMS will produce the same power into a given impedance that 10 Volts DC would produce (onto a resistance) Any waveform of 10 V R.M.S.will produce the same power into that impedance. This is because it's the root of the mean of all the average squared voltages to which Norbert Hahn referred in the prior post. It is if little meaning to compute the mean of squares of all the power values in a wave. 
RMS, when applied to power measurements, has come to mean "sine-wave power." A 100 Watt "RMS" amplifier can produce a 100 Watt sine-wave into its load. With music, the total actual power would be less. With a square-wave, it would be more. 

DIN power
The DIN 45000 defines different methods to measure power, depending on the device under test. Well, this is what I remember from reading the DIN some 25 years ago. 
For home applicances there are three different numbers for power: Continous power, Peak power and power bandwidth; the latter does not apply for speakers. 
Power measurement of an amp requires that the amp is properly terminated by Ohmic resistances of nominal value both at input and output. The continous power is measured when the amp is supplied by its normal power supply. It must then be able to deliver the rated power at 1 kHz for at least 10 minutes while the maximum THD does not exceed 1 %. To measure the peak power the normal power supply is replaced by a regulated power supply and the time for delivering the power is reduced. Thus, higher values for peak power are obtained. You may skip measuring the peak power by simply multiplying the continuous power by 1.1. 
The power bandwidth is defined as the bw for which 1/2 of the rated continous power can be obtained. 
Actually, DIN 45 500, CNF 97-330, EIA RS-426 and the encompassing IEC 268-5 specify not pink noise, but pink noise filtered by a filter that provides sinificant attenuation in the low and high frequency region of the spectrum to more closely model the long-term spectral distribution of music. Pink noise itself does not accomplish this 

PMPO (Peak Music Power)
So called "music power". This power figure tells the power which the amplifier can maximally supply in some conditions. PMPO rating gives the highest measuring value, but this info is quite useless, because there is no exact standard how PMPO power should be measured. 
The reason for this power rating was to show the max capability of equippment for recreating strong musical tansients like kettle drums and the like. Similar thing (music power rating) was used in the sixties, and I think it assumed a square wave that swung the whole supply range of the output stage. This alone gives them a factor of two over a clean sine wave note. But the ugliest thing they did was to assume that the high power lasted such a short period of time that the power supply caps would hold the voltages steady without any drooping. In the real world, an under powered PS could be hidden by this ruse and the PMPO might be a factor of 10 or more higher than what could be sustained on a nice instrumental performance. 
Forget what adverts say about peak power or other "power terms" because they are not standardized and anyway comparable between equipments. Just look for "RMS continuous Power" or other reliable power rating (like DIN power). 

Speaker power ratings
The nominal power for speakers is defined quite differently: The continous power is measured by pink noise rather than a sinousoidal signal and it is applied for 24 hours. Bandwidth of the noise is as required/specified by the speaker. Thus the nominal power is applicable to both a single chassis/driver and complete box. And the THD is not the limiting factor: It is replaced by the term that the speaker should by no means be damaged. Rhe requirement is that the speaker meet the manufacturers performance sapecification after the power cycle. 
The maximum power is defined for woofers and boxes only. It is measured by applying sinusoidal signals of 250 Hz and lower such that the speaker is neither damaged nor produces unwanted output. 
The AES/ANSI spec provides for two power measurements: thermal power, as you describe above, and excursion limiting, which is determined by either the hard mechanical limits afforded by the suspension, or the difference between the length of the voice coil and the length of the magnetic gap. 

Other amplifier specifications

Speaker impedance the amplifier is designed to drive
Many amps manufactured these days are rated only for 8-ohm-and-above loads, and not for 4-ohm loads. This is done largely as a cost savings by the manufacturer. Amps which are capable of driving 4-ohm loads to the same output voltage require heftier power supplies, heatsinks, and (often) output-stage transistors: they'll be delivering twice as much current into the load, and will be dissipating roughly twice as much heat within their output stages. 
If a manufacturer chooses to quote a power rating at 4 ohms in their advertising, the amp must be capable of delivering this much power after a 'warmup' period of operation at 1/3 power (which level actually dissipates _more_ heat in the output stage than full-power operation). 
In order to save money during manufacture, manufacturers often use skimpier power supplies, heatsinks, and output stages - and as a result, the amps may have a 4-ohm power rating which is _less_ than the 8-ohm rating. This is somewhat embarrassing for the manufacturer to advertise - and, so, they often do not quote a 4-ohm power rating at all, and state that the amp is designed to be used only with loads of 8 ohms or above. 

With many such amplifiers, you can drive a 4-ohm load safely, as long as you don't try to drive it too hard. If you drive a low-Z load to too high a volume, one of several things may happen: the amp may begin to "clip" (sounds very harsh and distorted, may damage the tweeters), or may overheat and shut itself down, or may overheat and burn up (all the magic blue smoke leaks out). 
 

Methods for making 4 ohm speaker to appear as 8 ohm
Wire a 4-ohm power resistor (10-20 watt) in series with each 4-ohm speaker. This makes the system to be appear as 8 ohm load and is inexpensive. The cons are that the resistor wastes power, may cause frequency response go bad because speakers do not have constant resistance with frequency. When you play at high volumes the resistor may get hot and burn thing or itself. 
Using 4 ohm to 8 ohm matching transformer will not waste much power, but the transformer will be heavy, expensive and hard to find. Transformer has also problems in playing back lowest frequencies (saturation causes distortion in high levels) and in higher frequencies the inductance in the transformer will cause phase shifts. You can wire two 4-ohm speakers in series if you have two identical speakers. Problem is that if the speakers are not identical type the frequency response and power distributin will be uneven. Most "8-ohm" amplifiers can drive a 4-ohm or 6-ohm load as long as you don't try to get full power out of the amp (if you do, it may overheat and shut down). 
Buy yourself a decent power amplifier whose output stage and power supply are capable of handling a real honest low-impedance load. Good amplifier will be expensive but gives best sound quality and reliabity. 
Dampling factor
The output impedance of an amp should be extremely low. If it's .8 Ohms, then an 8-Ohm speaker has a damping factor of 10. If it's .08, then the amplifier provides a damping factor of 100, etc. Don't confuse the actual output (source) impedance with the load impedance that is recommended for the amp (4-Ohms, 8-Ohms, etc). 
The idea is that if the speaker is 8 Ohms, and the amplifier has a source impedance of .08 Ohms, then the amplifier "damps" the motion of the cone by a "factor" of 100. In reality, the true damping that the cone "sees" is determined by many things, part of which is the damping limitation imposed by the resistance of the voice coil, usually around 5 Ohms or so for an 8-Ohm speaker. You can see that if the speaker has 5 Ohms of resistance, the internal (source) impedance of the amplifier (.08 Ohms for a damping factor of only 100) doesn't add much to the total resistance in the voice coil circuit, hence has very little effect on total damping. So any modest change in the amplifier damping factor correlates to virtually no change in total damping. 
A speaker designer shoots for a certain damping (same as 1/Q) to achieve a certain desired type of low-frequency rolloff. The assumption is that the source impedance of the amplifier is 0 Ohms. If the source impedance is .08 Ohms (damping factor of 100), very little error is introduced into the system. Higher damping factors are getting into diminishing returns in terms of the total damping. In practice we want a certain, relatively low damping figure for the whole speaker system, (1.414 for a maximally flat bass response). 

What is amplifier "bridging" or "monoblocking"?
When you're told a stereo power amplifier can be bridged, that means that it has a provision (by some internal or external switch or jumper) to use its two channels together to make one mono amplifier with 3 to 4 times the power of each channel. This is also called "Monoblocking" and "Mono Bridging". 
Bridging typical HIFI amplifier involves connecting one side of the speaker to the output of one channel and the other side of the speaker to the output of the other channel. The channels are then configured to deliver the same output signal, but with one output the inverse of the other. The beauty of bridging is that it can apply twice the voltage to the speaker. Since power is equal to voltage squared divided by speaker impedance, combining two amplifiers into one can give four (not two) times the power. 
In practice, you don't always get 4 times as much power. This is because driving bridging makes one 8 ohm speaker appear like two 4 ohm speakers, one per channel. In other words, when you bridge, you get twice the voltage on the speaker, so the speakers draw twice the current from the amp. 
Another interesting consequence of bridging is that the amplifier damping factor is cut in half when you bridge. Generally, if you use an 8 ohm speaker, and the amplifier is a good amp for driving 4 ohm speakers, it will behave well bridging. 
Also consider amplifier output protection. Amps with simple power supply rail fusing are best for bridging. Amps that rely on output current limiting circuits to limit output current are likely to activate prematurely in bridge mode, and virtually every current limit circuit adds significant distortion when it kicks in. Remember bridging makes an 8 ohm load look like 4 ohms, a 4 ohm load look like 2 ohms, etc. 
If your amplifier does not have built-in bridging option built in you can use an additional stage to invert the signal for one channel but drives the other channel directly.
 

The Speakers 

The speakers are the output of the system they are very important.
The common spearkers ploblem is the bass reproduction.
Here are two factor that make the difrence.

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 maximing 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 milimeters. 

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. The first speaker is a JL Audio 10W6 with an Xmax of .468" (12 mm), the second is a real 10" woofer from a prominent car audio manufacturer with an Xmax of 0.25" (6.5 mm), which at the time of this writing is pretty average in the industry. Let's call the second speaker "Speaker A". 

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 
 10W6
(Xmax = 12 mm)
 Speaker A
(Xmax = 6.5 mm) 
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, the 10W6 outperforms Speaker A 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 A's to equal the output capability of one 10W6. That makes sense when you consider that the 10W6 is moving virtually twice as much air as one Speaker A. 
 
 

If you refer to the plot to the right (clicking on this image will download a full-size version) 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 the 10W6 handles twice the power and is easily capable of outperforming Speaker A in this real-world situation. You will also notice that the 10W6 does not begin to approach its excursion limits until the frequency drops below 25 Hz, whereas Speaker A 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 10W6 and Speaker A, you can also see that low-frequency power handling is directly linked to Xmax. The 10W6 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 A at 40 Hz it will begin to distort and could potentially be damaged. The 10W6 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. 

Shameless Plug 

All JL Audio subwoofers feature very long excursion capability. Even our least expensive subs are more excursion-capable than many of the "top of the line" subs on the market. As you move up to the W4 and W6 subwoofers, excursion capability increases as does thermal power handling. The long excursion capability of JL Audio subwoofers not only ensures superior output capability but also superior fidelity with demanding program material. When the voice coils of lesser subwoofers are playing leap-frog with the magnetic gap, the JL Audio subs are still operating well within their linear range and producing clean, high-fidelity bass output. 
 
 
 

Hope you like it, come back soon, e-mail me.