Cable 

When I started to design cables in 2001, the first question that I asked was: “What is a good cable?” On the surface, designing a cable is easy because they are the simplest component in an audio system. In its simplest form, a cable is no more than a pair of insulated conductors terminated by plugs. Such a device should be easy to manufacture and they should all sound alike. However, this is not true in our world. None of the numerous brands of audio cables sound the same. This means they change the signal one way or another. In fact, it is a common practice for audiophiles to adjust the sound of a system by switching cables.

 

Before starting to research how cables change the signal, we must understand what an audio signal is. There are two categories: analogue and digital signals.

 

Analogue Signals

The electricity coming out of the microphone during a recording session are analogue signals. When in use, the microphone’s diaphragm moves in unison with the sound waves generated by the music to convert them into AC signals which means they change polarity. Most musical signals are very complex blends of many frequencies. Although the frequency bandwidth audible to the human ear is from 20 Hz to 20,000 Hz, but quite a few of the musical instruments have overtones outside the audible bandwidth, ranging from subsonic (below 20 Hz) to super-sonic (above 20,000 Hz) frequencies. If we feed musical signals into an oscilloscope, we can see a very complex wave having sharp peaks and dips, which is constantly changing in accordance with how the music goes.

 

The initial signals from the microphone are first fed into various electronic instruments for processing and are finally stored either  magnetically on tapes, or as mechanical undulations in vinyl records. During the reproduction process, the magnetic or mechanical signals are converted back into electricity. These electric signals are amplified through various electronic stages until they reach the outputs of the power amplifier at a strength strong enough to drive the speakers.

Digital Signals 

 

Nowadays, the analogue signal generated by the microphone is usually stored digitally by using a machine called the AD converter (analogue to digital converter). Its job is to convert the continuous analogue signals into pulses which are stored on magnetic tapes. In the digital domain, the signal is expressed in two forms: 1 and 0.1 is a sharp voltage pulse; 0 is no pulse. The analogue signal is modulated inside the stream of 0 and 1. Digital signals exist in all the digital components of the audio system, including CD machines, DA converters, digital preamplifiers and digital power amplifiers.

The common feature of both the analogue and digital signal is its dynamic, rapid changing nature. In the analogue domain, it is a complex, constantly changing AC signal. Digital signals are a series of rapid pulses.

 

Hence, accurate transmission is preserving the signal's exact order of changes with respect to time; in other words, retaining its wave form fidelity. The absolute signal strength is not important since we are constantly turning the preamp's volume control to obtain a comfortable listening level. 

The next important question was: "What physical phenomena are responsible for signal alternations in a cable?" There are five:
A) capacitance,
B) resistance,
C) inductance,
D) multi-pathway time difference,

E) external interferences.

A) Capacitance  

 

When two conductors are placed close together and are separated by an insulator (the dielectric), a capacitor is formed. This means every cable is a capacitor which must has a certain amount of capacitance. Capacitors are energy storage devices, accumulating electric charges on the surface of the conductors. These charges are released when the polarity is changed. Capacitors are widely used in power supplies to smooth out ripples in the current. For a cable, it is all bad news. Capacitance evens out the peaks and dips of the signal, and introduces time distortions by delaying the time interval between the signal changes.

Capacitance cannot be eliminated from a cable. It can only be minimized by spacing the conductors as far apart as possible, and also by the employment of clever conductor geometry.

The dielectric of both the audio capacitors and audio cables is a very important component because it affects both their speed and frequency response in a big way.

B) Resistance

Cable runs are short in home audio, no more than a few meters in most cases. Cable resistance is not high enough to affect sound reproduction. 

C) Inductance

Inductance is the occurrence of a secondary current (induced current) when an electric signal is traveling through a conductor. The phenomenon was discovered and termed self-inductance by Michael Faraday in 1831.  

An electric current has two components. The electric part is the current which is made up of the stream of electrons travelling between the sending and receiving ends of a conductor. The other component is magnetic. Faraday envisages it as an invisible field emanating from the conductor in all directions. The stronger is the magnetic field, the more distance it can radiate out from the conductor. The induced current is weaker in strength than the primary current, and travels in the opposite direction.

When another conductor is located inside the magnetic field generated by the primary current, a current is induced onto the second conductor. This is called mutual inductance.

In a cable, the minimum number of conductors necessary to complete the electric circuit is two. Therefore, there are both self and mutual inductance in every cable. These self and mutually induced currents generate a series of other currents of decreasing strength and different polarities in an array of subsequent time intervals, much like the echoes of sound waves generated by reflecting surfaces.

Inductance also evens out the dips and peaks of the signal, and opposes its polarity changes. The net effect is total chaos in the time domain.

 

Group Delay

In an electric signal, inductance produces a resistance gradient which increases with the frequency. Having the least resistance in the audible spectrum, low frequencies travel the fastest, and are capable of penetrating deep into the conductor. On the contrary, the slower high frequencies (possessing higher resistance) could only ride on the conductor's surface. The result is that different frequencies arriving at the other end of the cable at different times. It is called "group delay", a very common form of time distortion occurring in audio cables.

 

It is easy to understand that inductance is the number one obstacle in achieving accurate signal transmission. Like capacitance, inductance cannot be eliminated from a cable, but can be reduced by clever conductor geometry.

D) Multi-pathway time difference

Audio cables come in three major design groups, consisting of solid core, multi-stranded, and Litz wire. The solid core design has a single conductor per polarity. Multi-stranded cables are lamp cords made up of a large number of non-insulated conductors. Litz wires have a multitude of individually insulated conductors for each polarity.

Hence, only the solid core design providing a single signal pathway can achieve accurate signal transmission.

 

E) External Interferences

 

It is needless to mention that our atmosphere is jam-packed with EMI radiations resulting from human activities which are increasing rapidly year by year. They must not be allowed to enter a cables, or else there will be no accurate signal transmission. These radiations react and change the audio signals permanently. No device can take them out once that had happened. 

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The Ideal Cable

 

A good cable is one having a very fast transient response (at least as fast as the signal swing), wide bandwidth, and minimum time smear. In order to possess all these virtues, there must be a drastic reduction in capacitance and inductance, the use of a solid core design, and proper dielectric. Right from day one, all my cables  have an elaborate shield which is grounded.  Grounding drains away all the garbage currents when EMI interferences hit the shield. This simple procewdure increases the performance of all my products for at least 30%.

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Patent Tecnology

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After much research into audio cable construction and soul searching, the moment of epiphany came when I figured out an unique conductor geometry that could drastically reduce the inductance of a cable carrying electric cable, may it be digital or analogue. The design is based on magnetic field cancellation. I filed for a US patent, which was granted to me in 2007. ​By virtue of its design, my technology works on nearly all electric devices, such as speakers, loud speakers, electronics circuits, as well as projectors and TV sets.​​

Speaker 

I started to design speakers right after CES of 2007. On the surface, a speaker is not that complex, consisting of a few drivers, a crossover, and an enclosure. But when one starts designing it, he or she will soon realize the speaker is the most complex component of an audio system.

Loudspeakers are transducers, converting electrical energy into mechanical motions, which in turn move the air to produce sound waves. Three forms of energy are involved in its operation: electricity/magnetism, mechanical, and sound. Law of physics says that during the conversion of one form of energy into another, huge losses and distortions occur. Speakers produce the most distortion in an audio system. They are also very inefficient. The average efficiency is 86 db/watt, which is  less than 1% efficient in converting electrical energy into sound.
 

Throughout the long history of speakers, their design has always been a half science, half art endeavor. There are formulas available to describe what is going on inside certain parts of the speaker that are based on lab conditions very different from those existing at our homes. Thiele and small parameters, the first accurate mathematical description of how a woofer behaves inside an enclosure, only appeared only in the 1970s.

In a typical listening room, almost 50% of the sound reaching the listeners ears is sound reflected from the internal surfaces. Having to take reflected sound into consideration is a unique feature of speaker design. As acoustics differ from room to room, nearly all speakers are measured in an anechoic chamber.

Now comes the million dollars question: "How would this so-called accurately measured speaker sounds in a normal room?" The audio engineer who designs the speaker has no answer. The really difficult part of speaker design is how to correlate measured data with audio performance; an accomplishment that very few designers are really good at.

In theory, the perfect speaker is the one capable of reproducing the full audio spectrum of 20 Hz to 20,000 Hz accurately at a realistic sound pressure level (abbreviated as SPL) by means of a single driver, avoiding all the distortions introduced by driver differences and the crossover. Nevertheless, such a driver does not exist in the real world because the parameters for reproducing the various frequencies in a driver are contradictory ... large and heavy membranes for low frequencies; smaller, lighter ones for the mid and high frequencies.

The only solution to solve the problem is to split the audio spectrum into sections so that each bandwidth is handled by an optimized driver.  A 2-way system is the minimum, while a 3-way design is more easy on the drivers at the frequency extremes.

 

Woofer

In order to produce low frequencies at a realistic level, a woofer has to move a lot of air. This means it must possess large moving parts. In most design, it is made into a cone to reduce distortions generated by membrane breakup. The amount of air movement is determined by the cone size and its excursion distance. In order to produce the same SPL, a small woofer, such as a 6” (15.24 cm) unit, the cone will have to travel a much longer distance than that of a 15” (38 cm) woofer.

The popular, so-called “long throw” woofers are equipped with a wide and loose rubber surround to accommodate long cone excursions. The spider (a structure near the voice coil which serves as a centering device) has to be loosely constructed to facilitate long excursions.  Although such “long  throw” woofers could produce deep bass out of a small diameter cone but they lack accuracy because the cone overshoots every time when it moves.  Contrary to what most people believe, the main restoring force to a woofer cone is provided by the surround and the spider, not the damping force on the voice coil coming from the magnet. On the other hand, a 15” (38 cm) woofer cone will have to travel a much shorter distance to produce the same SPL. Hence, the surround and spider of the 15" driver can be made rigid to achieve a much more precise control on cone movements.​ It is also possible to build a large voice coil for big woofers to provide much better interactions with the magnetic field, resulting in more accurate translation of the electric signals into mechanical movements.  Another bonus of a large voice coil is an increase in efficiency. While the average efficiency of a 6” woofer is 84-86 dB/watt, it is 95-97 dB/watt for a 15" woofer. Please remember 3 dB means a power difference of 100%.

How the woofer is loaded is as important as its design, which is the key factor to determine the size and shape of a dynamic loudspeaker.

 

A woofer radiates sound waves on both sides of the membrane when in motion. The front and back radiations must be separated to avoid severe cancellations.

The IB (infinite baffle) solution is to mount the woofer on a very large sealed enclosure so that the air inside will not interfere with cone movements. For home applications, this means an impractically large enclosure.
   
The horn loaded systems are enormous, and even the folded horn speakers are huge. The price for increased efficiency is a lack of deep bass. High distortions are also generated when the sound waves travel through convoluted tunnels inside the speaker cabinet.

 

Bass reflex speakers have a rather small perforated enclosure designed to invert the phase of the back sound waves 180 degrees to become in phase with the front radiations. It offers the best compromise between size, efficiency, deep bass production, and high SPL capability. It is for this reason that I chose bass reflex loading in all my speaker systems. ​

 

Midrange Driver & Tweeter

The most common midrange driver employed in a 3-way system is a 3–4” (7.62- 10.10 cm) cone driver, operating at the 300 Hz–3,000 Hz range. The major problem of this configuration is that the crossover points are right inside the most sensitive frequency range of our ears. 

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Sound waves of various frequencies travel through air differently. Low frequencies have a spherical, omni-diredtional dispersion pattern. As the frequency increases, the waves propages at an increasingly narrower angle to become a narrow beam which are only audible on the tweeter's axis.

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The dome tweeter, having a far better dispersion than a cone tweeter by virture of its physicsal shape, has dominated tweeter design since the early 1950s. Most modern cone midrange drivers also have a dome-shaped dust cover over the voice coil built for the same purpose. 

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Because the midrange drivers and dome tweeters radiate directly, the membrane has to produce the full SPL of the music input.  The high excursions generate distortions and limit their efficiency. The maximum sensitivity of a 1” (2.54 mm) dome tweeter is no more than 93 dB/watt.

 

Compression Driver

The compression driver is a clever device dating back to a century ago. It has a very small, almost completely sealed chamber surrounded by powerful magnets. Inside the compression chamber is a dome-shaped membrane with a very large voice coil built into its surround. The vibrations of the membrane compress the air which exits into the atmosphere through tiny narrow slides. Amplification of the signal occurs during the transition from a high-pressure low particle velocity status inside the driver into a low-pressure high particle velocity condition when the air exits the compression chamber. 

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The compression driver is usually coupled to a horn which amplifies the sound for another 5 – 6 dB mechanically. A modern compression driver/horn assembly has an efficiency in excess of 110 dB/watt. Compression drivers are low in distortion and fast in speed by virtual of its ingenious design—the membrane has to move a very short distances to produce a high SPL, most of which is achieved through mechanical amplification. To enhance performance and to reduce distortions, the membrane is made of light weight but stiff metals, such as aluminium or beryllium.  Another of its benefits is the wide frequency range, typically 500 Hz–20,000 Hz for modern designs. This means the crossover frequencies can be moved away from our ear's most sensitive frequencies.

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Crossover

 

The other major component of a dynamic speaker is the crossover. Its job is to divide the audio spectrum into sections, and to balance the efficiency of the drivers.

 

Nearly all crossover networks use capacitors and inductors for frequency attenuation to enhance driver performance. A woofer has to have frequencies higher than its crossover point attenuated to reduce interference with the midrange and the tweeter. High levels of low frequencies fed into compression drivers overload and may even destroy them.

According to its design, a filtering network has a certain attenuation rate called the filter slope, providing 6 dB,18 dB, 24 dB, 36 dB, or even 48dB attenuation per octave. The steeper the slope, the more time delay occurs, which is expressed mathematically as a phase shift angle. For the best driver integration, the dividing networks should have the same phase angle. It is a difficult job for the engineer because different drivers need different filter slopes for the best protection.

Woofers are usually less efficient than the midrange units or the compression drivers. An attenuating network, usually in the form of an L-pad, is employed to reduce the outputs of the more efficient drivers.

Woofers, due to their larger size and heavier moving elements, have a much slower response time than tweeters and midrange drivers. Some crossovers contain time delay circuits to compensate for the differences in response time between drivers. But this electronic solution introduces distortions of its own. The problem is best dealt with by aligning the drivers on different vertical planes to ensure all sound waves will arrive at the listening position simultaneously.

Which part of the driver should be chosen as the reference point? Most audio engineers make the mistake of aligning the voice coil that does not generate any sound. The correct way is to align the acoustic center of the drivers.

The Ideal Design
 

1) Use the minimum number of ways to reduce distortions introduced by driver differences and multiple crossovers. The basis of all Orinda Acoustics designs is the 2-way system having a crossover point very close to 1,000 Hz. All our 3-way systems, including the gigantic 16” 3-Way and the brand-new flagship 18" 3-Way, are essentially an augmented 2-way design. The tweeter cuts in at above 10,000 Hz.

2) One driver per way to avoid the time distortion introduced by multiple drivers.

3) Design the simplest crossover using the least number of the highest quality components. Filter slopes are chosen to achieve a phase coherent configuration for all drivers.

4) Drivers are aligned on different vertical planes instead of time delay circuits.

 

5) The crossover is full shielded and grounded to provide the best signal protection. 

Amplifier Technology

I have been using tube gear when I was a hobbyist. My favorites were triode single ended 300B and 845 power amplifiers. These were used to drive my TAD 2401 Twin studio monitor speakers for many years. They were fine until I started using them to drive my speakers. The TAD has an efficiency of 98 dB/watt while those of mine has an efficiency ranging from 91 dB/watt to 96 dB/watt (the Sunny Cable Technology models). These triode amplifiers just do not have enough power to drive them. Besides high distortions, there is a lack of control in the bass resulting from the small damping factor common to all single ended valve amplifiers. The situation was a lot worse in the home theater system ... explosions and earthquakes were not realistic. I had no option but to ditch my beloved triode amplifiers and switched over to solid state amplifiers. The amplifier that I was using while I was at Sunny Cable Technology was a modified 200 watts per channel FET amplifier made by Ayre. However, I was never completely satisfied with its performance.

I did not stopped researching on the transistor amplifier even after I retired in 2009. My requirements were:

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1) It must have sufficient output to drive my speakers without distortion. At that time, my least efficient speaker was 93 dB/watt. An amplifier having an output of 60 watt at 8 ohms is sufficient.

 

2) It must be lightweight and compact so that carrying the amplifier around is easy.

 

3) It should emit little heat during operation, a class AB design which could be left on indefinitely is ideal.

After much searching, I found the perfect circuit; neutral, clean, yet without a hint of being solid state.

Work on the preamplifier did not go that well. Most of them are colored and veiled. To my surprise, the cleanest sounding preamp is a passive one, which is essentially a stepped volume control.

 

My amplifier is also equipped with a unique dual chassis, based on my patent technology to facilitate the most faithful signal processing in a noise free environment.

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AC Power Conditioner

 

My first AC conditioner was termed "THE BOX", which made its first appearance in the 2005 Las Vegas CES. Its predecessor (nicked-named "Anaconda") was a 20 ft long (6m), 4" (10 cm) wide, 70lbs (31.75 kg) giant cable, that premiered at our first CES show in the previous year. 

 

"THE BOX" is a passive device, containing no active components, such as transformers, capacitors, inductors and electronic circuits. It is essentially a very long cable made using our patent technology. The beauty of my invention is that a sufficient length of it cancels out most of the noises from the environment which were picked up by the long cable runs from the AC generator to our wall outlets.  

 

Another major function of "THE BOX" is to provide a lag free, instantaneous current supply to the electronic circuits. Some explanation is necessary for audiophiles to understand how it works. The AC power all over the world has a fixed voltage. For example, 110 volts in the USA and Canada, 220-250s volts for most European countries. Since the voltage is fixed, and voltage x current = power; the amount of electric energy coming out from the AC wall outlet is directly proportional to the current being drawn. Solid state circuits (those found inside preamp, power amp, D/A converters, and mixing consoles) draw current according to their workload. Instantaneous current supply is the key factor for all of them to work well. Active power conditions are slow and sluggish in comparison, starving our beloved power amps of the vital current necessary to drive the loudspeakers for faithful music reproduction.

 

The same double chassis technology utilized in my amps is also included in ''THE BOX". 

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