There is a law in the TALMUD, (the ancient studies of Jewish law and philosophy ) which I feel should be widely applied today; "A person is not permitted to ask the storekeeper for the price of an item if he knows that he will not purchase it."
The information I want to empart to you, is meant to enhance your understanding of how your system works. Not to impress upon a working person whose time is precious. A missed sale because you occupied his/her time while other customers are there, knowing you do not intend to buy NOW, is robbing that person of a meager income that we all complain about. Think twice next time you have a whim to take a test drive in a new model car just... because! Would you have it done to you?
One more NOTE* ( the most important one of all ) The most avid audiophile will not bother with specification sheets as much as the RESULTING sound. Remember you are shopping for sound with your ears, not eyes . Specification sheets will NEVER tell you how any unit will sound like with your equipment in your surroundings. These specifications are derived under laboratory conditions, not actual use. Trust YOUR ears, no one else's.
 | AUDIO-FILE |
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- Before I jump to the next item, I would like to emphasize the order of importance for selecting an audio system. The most important of which is the loudspeaker, being your source of information. Next would be the system's source, such as the Compact Disc, or Turntable, Cassette, VCR, etc… This is the Amplifier's source of information. The last component is the amplifier, It is the determining factor that will make or break your system.
- We must understand what sound is. In plain terms, sound is air in motion. - Since we are all fameliar with the basics of a recording process, as in an orchestra that was recorded at a venue such as Carnagie-Hall, the sound was picked up by microphones, and transferred eventually to a playback medium such as a C.D. or Tape, and vinyl records. - In order to understand the basics of how a speaker or microphone works, we need to remember some more science basics, relating to magnetism.
- The basics that we need to know here will cover some questions about the workings of a speaker, and a microphone, which in essence are the same, as you may soon learn.
- Remember that if we take a thin copper wire, and make it into a coil, then attach a diaphragm to it, such as a thin membrane like cellophane, and insert the coil into a magnetic field, where the magnetic field is strongest, and cause the diaphragm to move, we will have generated an electrical current across the two leads from the coil. This is a basic description of a microphone , or speaker.
Our voice, or music from an instrument, moves air molecules that are picked up as vibrations by the diaphragm. Causing the attached coil to move back and forth in the magnetic field, creating an electrical charge through the copper coil, which an amplifier picks up, makes it larger and stronger, so it can move a heavier diaphragm, like the speaker cone. Which weighs considerably more, and needs to move more air, to reproduce the volume level we need.
- The basic components of a speaker consist of, The cabinet, The filtering system known as a dividing network, Or Crossover, Which consists of capacitors and coils. These components send the necessary tones/frequencies to the individual speakers, such as the woofer and tweeter, that are mounted in the cabinet.
There are several types of loudspeaker designs to choose from, such as;
Orthodynamic/Electro-Magnetic being a standard type of speaker.
The Ribbon speaker.
The Electrostatic type.
The designs for the Orthodynamic/Electro-Magnetic type consist of three basic designs. The full range type, where only one speaker is used to reproduce the full audio range. The two-way type, using a dedicated speaker for bass and midrange, called a woofer, and a smaller speaker designed to reproduce treble sound, known as a tweeter. A 3-way design uses the addition of a medium sized speaker to reproduce mid-range tones such as voice, and horns.
The Ribbon type, uses a foil-like material, which is statically charged, and moved by changing the magnetic field of the magnets which move the foil.
The Electrostatic type,uses a diferent material, which can also be staticaly charged, and uses the same principal as the ribbon type.
Here is another point to consider. The lower the tone of sound, The more it will radiate naturally, Just as when you hear the famous " Boom-Cars " that have surrealistic loud sound, The Bass is heard or felt long before the upper tones such as treble. The higher the tone the more directional it becomes, And we rely on treble information to recreate detail, sound ambience, and proper staging, separation etc. Typically the ribbon and the electrostatic type , are not capable of reproducing the lower bass notes very well, so an orthodynamic bass woofer may be added in some cases, making a hybrid design.
When it comes to selecting a speaker that will suit your requirements, and since the speaker placement is crucial, and should not be placed in the corner of the room, or behind anything , you need to answer some basic questions, such as:
How often will you plan to use it, serious listening, or passive?
Is the size of the speaker important in relation to its placement location?
How big is the room? If you have an adjoining room, a bigger speaker, or the addition of a subwoofer may be necessary, and possibly a more powerful amplifier required. And consider open living rooms to include all adjoining rooms and hallways.
Are the esthetics of the speaker important? In order to match the room's décor.
Does it sound pleasant ? Can you listen to it for a long time without getting irritated ?
Will it operate well with the amplifier?
Be prepared to allow a majority of the budget towards the speakers.
In some cases, a sub-woofer may be desirable to augment a systems performance, by reproducing the lower bass notes such as the percussion of a harp, or the depth of low bass from a large pipe organ , electric bass.
The addition of a sub-woofer will also help by providing depth of bass at lower volume levels, where the main speakers may be incapable of moving the mass of air required.
This addition I find helpful, in reducing the potential to damage speakers, by providing depth of bass with little effort. Something we all want, which is why we turn UP the volume.
Yet another reprint from:
QUAD the British manufacturer of fine audio electronics.
Electrostatic loudspeakers have inherent advantages over conventional loudspeakers. An electrostatic loudspeaker consists of a very thin plastic membrane suspended between a pair of conducting perforated plates. The membrane carries an electrostatic charge and is forced to move by the electrostatic field produced between the plates when the signal voltage is placed across them. The membrane is very light (the thickness is one tenth that of a human hair in the
Thus the two major causes of coloration in *(ElectroMagnetic/Orthodynamic) moving coil loudspeakers, which we have come to regard as the sound of hi-fi, are avoided in electrostatic loudspeakers. The performance of the ESL 63 is a revelation to anybody accustomed to listening through moving coil loudspeakers.
The *(ElectroStatic Loudspeakers ) "Quad ESL 63" takes the performance advantages of electrostatic loudspeakers a stage further. The ideal loudspeaker for stereo reproduction is a single point source reproducing all frequencies. Loudspeaker engineers have been trying to develop one for more than half a century. If it is not possible to produce a point source loudspeaker, can we instead make a loudspeaker that to an observer behaves like one?
*(TECHNO stuff )
Imagine a theoretically ideal point source loudspeaker radiating sound pressure waves and then imagine a plane in the air a short distance from the source and at right angles to the direction of propagation. If the air at the plane is made visible in some magical way, we will see concentric waves radiating out from the centre just as they do when a stone is thrown into a still pool. If we substitute an electrostatic loudspeaker membrane for the plane in the air, make it move in exactly the same way as the air on our imaginary plane and suppress the imaginary source, the results to an observer positioned on the far side will be identical to those from the ideal source. The *( ElectroStatic Loudspeaker ) "Quad ESL 63" does exactly this.
An ingenious arrangement of concentric electrodes fed by a sequential delay line produces a sound pressure pattern that is an exact replica of that from an ideal source placed 30cms behind the plane of the diaphragm. The *( ElectroStatic Loudspeaker ) ESL 63 is a totally homogenous sound source, phase true and aperiodic, with a frequency response, both on and off axis, quite free from the irregularities that are inevitable with any multi-way loudspeaker system. It has a very well controlled directivity characteristic with the result that there is no stereo hot-spot.
A pair of 63s *( Electrostatic speakers ) will produce an excellent stereo image over a range of listening positions, which are as wide as the speakers are apart. Since the loudspeakers can be placed right up to the side walls the stereo stage can be very wide and the listening position very free. With the very best stereo recordings, the results are holographic, and moving from one side to the other presents the orchestra from different points of view.
*NOTE : I have placed this article for your information only , this is not an advertisment nor suggestion . Although I consider " QUAD " to be an excellent choice for Hi-Fi , it may or not suit your needs .
Most audiophiles will opt for separate components when selecting their system. The idea here is that each component will have it's own power supply, and reduced interferance from adjacent circuitry, along with better quality circuit components.
The following section is a re-print from a page on the web, explaining the different types of amplifier designs common to the Audio industry.
Audio power amplifiers are classified according to the relationship between the output voltage swing and the input voltage swing of a given sine wave signal, thus it is primarily the design of the output stage that defines each class. Classification is based on the amount of time the output devices operate during one complete cycle of signal swing. This is also defined in terms of output bias current [the amount of current flowing in the output devices with no applied signal]. For discussion purposes (with the exception of class A), assume a simple output stage consisting of two complementary devices (one positive polarity and one negative polarity) -- tubes (valves) or any type of transistor (bipolar, MOSFET, JFET, IGFET, IGBT, etc.). *added by E.C *(Field Effect Transistors) I.E : Metal Oxide Semi-conductor Field Effect Transistor = M.O.S.F.E.T
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Class A operation is where both devices conduct continuously for the entire cycle of signal swing, or the bias current flows in the output devices at all times. The key ingredient of class A operation is that both devices are always on. There is no condition where one or the other is turned off. Because of this, class A amplifiers in reality are not complementary designs. They are single-ended designs with only one type polarity output devices. They may have "bottom side" transistors but these are operated as fixed current sources, not amplifying devices. Consequently class A is the most inefficient of all power amplifier designs, averaging only around 20% (meaning you draw about 5 times as much power from the source as you deliver to the load!) Thus class A amplifiers are large, heavy and run very hot. All this is due to the amplifier constantly operating at full power. The positive effect of all this is that class A designs are inherently the most linear, with the least amount of distortion. [Much mystique and confusion surrounds the term class A. Many mistakenly think it means circuitry comprised of discrete components (as opposed to integrated circuits). Such is not the case. A great many integrated circuits incorporate class A designs, while just as many discrete component circuits do not use class A designs.]
Class B operation is the opposite of class A. Both output devices are never allowed to be on at the same time, or the bias is set so that current flow in a specific output device is zero when not stimulated with an input signal, i.e., the current in a specific output flows for one half cycle. Thus each output device is on for exactly one half of a complete sinusoidal signal cycle. Due to this operation, class B designs show high efficiency but poor linearity around the crossover region. This is d ue to the time it takes to turn one device off and the other device on, which translates into extreme crossover distortion. Thus restricting class B designs to power consumption critical applications, e.g., battery operated equipment, such as 2-way radio and other communications audio.
Class AB operation is the intermediate case. Here both devices are allowed to be on at the same time (like in class A), but just barely. The output bias is set so that current flows in a specific output device appreciably more than a half cycle but less than the entire cycle. That is, only a small amount of current is allowed to flow through both devices, unlike the complete load current of class A designs, but enough to keep each device operating so they respond instantly to input voltage demand s. Thus the inherent non-linearity of class B designs is eliminated, without the gross inefficiencies of the class A design. It is this combination of good efficiency (around 50%) with excellent linearity that makes class AB the most popular audio amplifier design.
Class AB plus B design involves two pairs of output devices: one pair operates class AB while the other (slave) pair operates class B.
Class C use is restricted to the broadcast industry for radio frequency (RF) transmission. Its operation is characterized by turning on one device at a time for less than one half cycle. In essence, each output device is pulsed-on for some percentage of the half cycle, instead of operating continuously for the entire half cycle. This makes for an extremely efficient design capable of enormous output power. It is the magic of RF tuned circuits (flywheel effect) that overcomes the distortion create d by class C pulsed operation.
Class D operation is switching, hence the term switching power amplifier. Here the output devices are rapidly switched on and off at least twice for each cycle (Sampling Theorem). Theoretically since the output devices are either completely on or completely off they do not dissipate any power. If a device is on there is a large amount of current flowing through it, but all the voltage is across the load, so the power dissipated by the device is zero (found by multiplying the voltage across the device [zero] times the current flowing through the device [big], so 0 x big = 0); and when the device is off, the voltage is large, but the current is zero so you get the same answer. Consequently class D operation is theoretically 100% efficient, but this requires zero on-impedance switches with infinitely fast switching times -- a product we're still waiting for; meanwhile designs do exist with true efficiencies approaching 90%. [Historical note: the original use of the term "Class D" referred to switching amplifiers that employed a resonant circuit at the output to remove the harmonics of the switching frequency. Todays use is much closer to the original "Class S" designs.
Class E operation involves amplifiers designed for rectangular input pulses, not sinusoidal audio waveforms. The output load is a tuned circuit, with the output voltage resembling a damped single pulse. Normally Class E employs a single transistor driven to act as a switch.
The following terms, while generally agreed upon, are not considered "official" classifications
Class F Also known by such terms as "biharmonic," "polyharmonic," "Class DC," "single-ended Class D," "High-efficiency Class C," and "multiresonator." Another example of a tuned power amplifier, whereby the load is a tuned resonant circuit. One of the differences here is the circuit is tuned for one or more harmonic frequencies as well as the carrier frequency.
Class G operation involves changing the power supply voltage from a lower level to a higher level when larger output swings are required. There have been several ways to do this. The simplest involves a single class AB output stage that is connected to two power supply rails by a diode, or a transistor switch. The design is such that for most musical program material, the output stage is connected to the lower supply voltage, and automatically switches to the higher rails for large signal peaks [ thus the nickname rail-switcher]. Another approach uses two class AB output stages, each connected to a different power supply voltage, with the magnitude of the input signal determining the signal path. Using two power supplies improves efficiency enough to allow significantly more power for a given size and weight. Class G is becoming common for pro audio designs. [Historical note: HITACHI is credited with pioneering class G designs with their 1977 Dynaharmony HMA 8300 power amplifier]. *(And the introduction of the M.O.S.F.E.T transistor, added by E.C.)
Class H operation takes the class G design one step further and actually modulates the higher power supply voltage by the input signal. This allows the power supply to track the audio input and provide just enough voltage for optimum operation of the output devices [thus the nickname rail-tracker or tracking power amplifier]. The efficiency of class H is comparable to class G designs. [Historical note: Soundcraftsmen is credited with pioneering class H designs with their 1977 Vari-proportional MA5002 power amplifier.]
Class S First invented in 1932, this technique is used for both amplification and amplitude modulation. Similar to Class D except the rectangular PWM ( Pulse Width Modulation ) voltage waveform is applied to a low-pass filter that allows only the slowly varying dc or average voltage component to appear across the load. Essentially this is what is termed "Class D" today.
CLASS-T The underlying technology of Class-T does not use Pulse-Width-Modulation and is not a pure analog approach ( such as Classes -A and -AB). It combines the benefits of both with a completely new approach which is DPP(Digital Power Processing) based in design. Class-T amplifiers provide the signal fidelity of discrete-component linear Class-A and AB designs, while offering high power efficiency and the potential to need simpler engineering, which reduces manufacturing cost for amplifier overhead at high power levels (such as power supply, filtering, and heat venting),in order to achieve high-end audiophile grade sound performance. Class-T claims to provide power conversion efficiencies of 80 percent to more than 90 percent, which is equal to or better than "Class-D" amplifiers. Class-T technology uses both analog circuitry and Digital Power Processing algorithms that modulate the input signal with a high-frequency switching pattern. Tripath’s proprietary algorithms are derivatives of adaptive and predictive algorithms used in telecommunications processors. The modulated signal is sent to output transistors then through a low-pass filter (external to the Tripath amplifier) that demodulates it to recover an amplified version of the audio input. In a Tripath amplifier there is an input stage that provides analog input signal buffering. The output of this stage drives the Digital Power Processing TM block. This block contains an adaptive signal conditioning processor, a digital conversion function, mute control, overload handling, fault detection, predictive processing and qualification logic functions. The output of the DPP TM block controls a power output stage that drives a speaker through an output filter. In a traditional Class-D PWM amplifier the audio input signal is compared to a higher-than-audio frequency (generally 100-200 kHz) triangle wave. The resultant signal drives switching transistors in a push-pull fashion. Amplification is achieved with the higher voltage and current that the output transistors switch to the speakers. A low-pass filter (inductor and capacitor) positioned before the speaker removes the triangle wave base frequency, leaving amplified audio.
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