Which advantages do balanced input signals offer

In contrast to unbalanced signals, balanced signals are carried by two wires (plus ground/shield). In the transmitting device, a balanced signal is created by generating an inverted original signal (180ˆphase shifted). The "hot" wire carries the original signal (a), the "cold" wire the inverted signal (-a). In the receiving device, the balanced signal is processed by a differential amplifier, which detects the difference between both: (a) – (-a) = 2a. On its way between devices, the useful signal can be affected by interference (s). Interferences however are in phase on both wires and fed to the differential amplifier as well. Again, the amplifier detects the difference between the interference contents: (s) – (s) = 0. Thus - in an ideal situation - all interference on the signal path is eliminated.

Why are discrete signal paths important

Twin op-amps are the most common design for operational amplifiers, i.e. two amplifier circuits are integrated in one device. If left- and right-channel signals are processed simultaneously by such a device, interaction between both cannot be excluded. This interaction is admittedly diminutive, but should be avoided whenever a different design offers the possibility.

Why are op-amps ideal for low-level signal processing

Discrete amplifiers (designed with transistors) are very popular in High-End audio design also for preamplifier stages. This is often marketed as an optimization measure, but the partially exorbitant extra expenses are of course to be paid by the customer.
But an op-amp consists of tranistors as well...
Moreover, its structure has the advantage of thermal coupling between its internal components.
Also ageing issues play a much less important role.
Due to the large number of op-amps types offered, it is possible to pick an optimum type for any specific application.

Why does frequency bandwidth limiting make sense

In signal processing, sound is represented by AC voltages. Sound is audible - for young people - from about 20 to 20000 Hz.
The elder the listener, the less he will hear high frequencies in particular.
In order to transmit these frequencies at optimum quality, the frequency response of an amplifier should be as wide and as "flat" as possible.
At the low end of the scale, this limit is represented by DC, as there is no frequency lower than zero.
In upward direction, the limit can be set to practically any frequency, but the higher, the more susceptible the device becomes concerning electro-magnetic interference.
This is not audible in the first place, but may interfere with the useful signal and then become evident.
Therefore, unrestricted frequency response attests thoughtlessness rather than remarkable engineering skill.

Why is a good volume pot essential

A volume potentiometer is a mechanical control element, which can be obtained on the market at any low price. Meanwhile it is often replaced by electronic circuitry, exhibiting essential disadvantages concerning dynamic range, noise and distortion. Conductive-plastic resistive tracks, high-quality multitap wipers and separated chambers for the individual sections are highly desirable for sophisticated applications, and high quality is inevitable to ensure trouble-free operation for years. Since the market for really good pots is a small one, manufacturers like Noble or Panasonic don't offer these any more. A current sample of top of the line pots is the RK27 by ALPS, which is used inside all Violectric headphone amps (except V90).

Digital Audio Converters

About digital volume control...

The benefit of digital volume control at first glance is, that there will be no more scratching, there will be no channel mismatch, there will be no crosstalk issues any more. Digital volume control can be made with up-down buttons or incrementals or real potentiometers – like Violectric does. In that case a linear tapper is used, because the volume control itself is made dB-linear inside the D/A converter. As this mirrors a “real life” feeling only imperfect when turning the potentiometer, we added some resistors to “bend” the responding law from the potentiometer to have a nearly perfect analog feeling. A simple DC voltage is attenuated by the pot. The result is fed to a A/D converter, here a digital control signal is made to attenuate the digital audio signal inside the D/A converter BEFORE converting it to analog. A digital 24 bit signal represents a dynamic range of 144 dB, a 32 bit signal represents a dynamic range of 192 dB. – much, much more than can be found in real life !!
A Vinyl record has its limits at 60 d, a tape recorder without noise reduction systems will not be better.
People who are doing real world recordings can tell, that it is nearly impossible to record more than 60 dB dynamic range with a microphone - although microphone makers claim dynamic ranges from their mics to be more that 130 dB. This may be true when recording a cricket near a starting F-14 Tomcat. But - who needs that. Also, sitting in you living room, it is hard to follow dynamic ranges of more that 20 – 30 dB unattanuated without having trouble with your neighbourhood afterwards.
Today´s pop music´s dynamic range is reduced during recording to 2 – 3 dB … Please also note that harmonic distortion inside the signal cannot be smaller than the dynamic range.
It is not possible to have 100 dB THD (0,001%) with 90 dB dynamic range,
but it is possible to have 110 dB THD (0,0003%) with 120 dB dynamic range ! The CD format offers 16 bit which means a dynamic range of 96 dB and distortions which cannot be lower than 0,0016%. A 24 bit signal offers a dynamic range of 144 dB with theoretical minimum distortions at 0,00001%. This is not possible to achieve in real life. This also concerns 32 bit signals.
The best today AD converters offer dynamic ranges from 120 dB with distortion figures about –110 dB THD. The limits of physics are not far away. Lots of losses have to be faced during recording, editing, mixing … so the CD as a final storage and play-back media is a totally sufficient solution. You cann´t figure much more with senseful efforts.
Digital attenuation is done by shifting the signal from MSB (Most-Significant-Bit) in direction LSB (Least-Significant-Bit). Shifting a complete bit in LSB direction (and replacing it with a 0) means 6 dB attenuation. Not only inside converters from Lake People and Violectric, 16 Bit signals from a CD are expanded to a 24 bit signal while passing the resampling unit or entering the D/A converter chip. Inside the new DAC V850 the digital signals are expanded to 32 bit signals. By doing so, at 16 bit signal may be attenuated inside a standard 24 bit converter by 6 dB x 8 bit = 48 dB = factor 200:1 WITHOUT changing anything from the original data. What will you hear in your living room when you attenuate the maximum volume to 0.5 % ! Inside a 32 bit converter the attenuation may be as much as 6 dB x 16 bit = 96 dB = factor 60.000:1 - far more than nothing ...
We learned from the above ( the "real-world" A/D converter) that also a real 24 bit signal carries a maximum of 20 “senseful” bits - in practice there are no more than 18 bits. So, also a 24 bit signal may be attenuated by a minimum of 6 dB x 4 Bit = 24 dB = factor 35:1 WITHOUT doing any harm to the original data. For a 32 bit converter the calculation is 6 dB x 12 bit = 72 dB = factor 4.000:1. So – for our opinion - digital attenuation is the best what can happen to a signal (except not being attenuated). Of course provisions should be made to adapt different working levels in the audio chain. It makes no sense to have a DA converter which offers its technical data only when dramatic amounts of output voltage is present on its outputs. And because you need only 1 or 2 of these volts you are forced to always digitally attenuate the signal in advance. The maximum output level from a DA converter should be adjusted in the analog make-up circuitry between DA chip and output sockets without changing output impedances and without using special measuring equipment. Afterwards, attenuating the signal digitally will not be an issue at all.

About resampling...

Resampling is a mighty feature for the transformation/restoration of jittered signals into high-quality signals. As well, the signal quality of 44.1 or 48 kHz sources can be improved by converting them to a higher rate. This also complies to the USB input which often suffers from jitter caused by improper treated computer outputs. With the aid of the resampling process nearly all jitter (and not only specific ones) is removed from all digital input data. This also concerns to the USB data stream without using “asynchronous USB mode” which often is the cause for the next issues. We all know that computers are solving problems which haven’t been existing before ... This isn't mystery at all, but a feature offered by so-called asynchronous sample rate converters - available since the nineties of the past century. While early sample rate converters could provide conversion ratios of 1:2 to 2:1 (at 100dB dynamic range) only, ratios of 1:16 to 16:1 at 140 dB dynamic range are feasible today.In principle, the digital data stream is asynchronously disassembled by a DSP specially developed for this purpose, and can be recombined at virtually any sample rate desired.By means of this process, all potential jitter vanishes almost completely and - due to the higher sampling rate - the analog filters after the converter stage can follow a much more straightforward and "musical" design. Furthermore, all digital input signals will be completed to 24 bit signals what is valuable in conjunction with the digital volume control.

About the analog output level...

Unlike in the analog world, digital technology uses a clearly defined maximum level, described as 0dBFs, or "zero deciBels Full scale" in clear text. From this maximum downward, signal levels are expressed with a negative sign. The “translation“ of the digital level into analog is provided by the D/A converter and is extremely flexible, whereby several standards have established.
Professional broadcast facilities in Germany - i.e. radio and TV stations - understand 0 dBFs as equivalent to 15 dBu analog level. In other countries this may be handled differently.
Notabene, 15 dBu represent a voltage of 4.5 Veff which may exceed the capabilities of many audio devices designed for a voltage swing of 1...2 volts. Therefore, the maximum output level of the DAC V800 can be adapted by means of internal DIP switches. While factory-preset to 15dBu, the maximum output level can as well be set to 24 / 18 / 15 / 12 / 6 dBu (with the volume level knob fully clockwise). The values above apply to the balanced outputs. Level at the unbalanced outputs will be 9 dB lower, i.e. 15 / 9 / 6 / 3 / -3 dB. Of course, all these settings are effected on the "active" side of the circuitry, thus leaving output impedance completely unaffected. By means of the active analog output adaptation to external conditions, there are only negligible effects on overall performance... if at all ! Furthermore, analog output level is controlled by means of a potentiometer on the front panel. Its setting is translated in the digital domain only, thus excluding any kind of scratching, crosstalk or gang errors. Since the potentiometer affects data word length (and so: signal accuracy), a setting closest possible to the right end stop should be chosen to achieve the maximum desired loudness. Coarse level should always be set by means of the DIP switches described above.

Headphone Amplifiers

Why it makes sense to make such huge efforts

A headphone amplifier is a device designed to condition audio signals with regard to the very specific requirements of headphones. This doesn't sound too spectacular at the first glance and can be achieved relatively easily. As with many things however, the devil is in the details and much more effort is required to design one amplifier for (nearly) all current headphone models. Headphones per se are quite diverse, and there are two essential parameters: impedance and sensitivity. In general, headphones with higher impedance can be regarded as less sensitive than headphones with low impedance (which is not generally true, but in the majority of cases). The sensitivity of headphones is usually stated in dB (sound pressure level) per Milliwatt. Extremes in this sense are the AKG K1000 with 74dB/mW on the one hand, and the Sennheiser HD25 with108 dB/mW on the other hand: The K1000 requires 2500 times the power to achieve the same sound pressure as the HD25. There is also the fact that headphones with high impedance usually require much higher voltage to achieve high loudness. Thus the amplifier must be designed with high internal supply voltages.

Why does an active feed-through make sense

Each electronic device presents input impedance as well as input capacitance. If several devices would be coupled passively - e.g. with "Y" adapters - the resulting input parameters could provoke malfunction and instabilities. A buffer amplifier "reconditions" the signal and makes it compatible with other devices due to its low-impedance output.

Why does PRE-GAIN make sense

Two extreme examples (with the headphone amplifier at 8dB (x 2.5) overall gain, volume control set to full): 1st example:
The (pre-)amplifier provides 4V output voltage, whereas the headphone requires only 2V for 100dB sound pressure level. With the control fully turned up, the amplifier would deliver 10V output at 8dB gain. Therefore the volume control would have to be operated very carefully in order to avoid hearing damage. Moreover, any interference at the input should be avoided since it would be "unforgivingly" amplified as well. With PRE-GAIN, the input level can be reduced by 12dB (a fourth), with 1V instead of 4V as the result. This 1V is again amplified by 2.5 (8 dB), then equalling 2.5V. Now the volume control can be turned over almost the entire range. 2nd example:
The (pre-)amplifier provides 1V, whereas the headphone requires 20V to release 100dB of sound pressure. With the volume control fully clockwise, the V200 would provide 2.5V at 8dB gain only - much to low for the headphone. By means of PRE-GAIN, input level can be boosted by 12dB (four times), resulting in effective 4V. These are again multiplied by 2.5 (8 dB), now equalling 10V. This is still not enough, but far closer to the optimum value: The headphone achieves 114dB sound pressure level.

Why are high supply voltages essential

A headphone doesn't really require high power, but from the equation P = U2 / R we can see that the square of the supply voltage determines the power into a given load resistance.
The higher the headphone's impedance, the more voltage will be needed.
But this deals with the achievable loudness to a limited extent only:
Technically spoken, music lives on fast transients which put high demands on signal processing.
And thus a fast transient can easily push an average amplifier with +/-15 volts supply to its limits.
Due to VIOLECTRIC´s high supply voltage you will benefit from doubled output swing capability.

Why is a low output impedance = high damping factor essential

When actuated, electro-dynamic systems respond with a counterforce.
When the voice coil of a headphone has been displaced by the signal, an (error)-current will be induced when it swings back to its initial position.
This current must be suppressed as far as possible, which is effected best if the amplifier's output impedance is the lowest possible.
The damping factor describes nothing but the ratio between output impedance of an amplifier and a given load.
Since there is no known technical specifications, we define the load (voice coil impedance) as 50 ohms.
This results in an output impedance of <0.06 ohms in case of the HPA V200.

Why does a relay make sense when switching power

Amplifiers generate unwanted output signals when applying or removing power, which can damage the connected headphones.
The relay breaks the connection between amplifier and headphone and thus protects the latter until electrical conditions have stabilized.

Phono Preamplifiers

Why turntables are susceptible to hum

Most turntables are connected unbalanced via cinch leads.
Unfortunately indeed, since a balanced connection would be hum-free in practice and much more advisable in technical terms.
Common pickup systems usually represent a perfect dipole, and thus the advantages of balanced signal connection could easily be made use of.
Moreover, vinyl recordings are being equalized during the production process, i.e. low frequencies are reduced and high frequencies boosted.
During playback, the equalizing preamplifier inverts this process, thus enhancing not only the low frequency range, but also any hum interference to the signal lead. When connecting the pickup in balanced mode, hum interference as would be unavoidable with unbalanced leads is significantly reduced.

Why did we choose instrumentation amplifiers for PPA V600

Like mentioned earlier, the balanced signals are fed to special instrumentation amplifiers with differential inputs, which were originally intended for the use with microphones.
Those instrumentation amplifiers are made to make lots of gain with only very small portions of noise - as low as no op-amp or a combination of op-amps could offer !
However, the difference between dynamic microphones and pickup systems isn't as big as one might assume:
By means of wire coils and magnetism, they both deliver very low signal voltage at quite similar impedance.
During design, we discovered that this breed of instrumentation amplifiers is ideally suited for turntable pickups as well.

Why an easily accessible gain control is useful

A further advantage of instrumentation amplifiers is that gain can be adjusted by rather simple measures.
Furthermore, gain control can be achieved without any significant change in bandwidth, as faced with (multi-stage) operational amplifiers.
Thus, bandwidth still exceeds 200kHz at a gain setting of remarkable 60dB, equalling a level ratio of 1:1000 !
By means of the front gain controls, the PPA V600 can easily be adjusted even for less known pickup models without hassle. Although there is only one kopb to be operated, gain setting is provided for both channels individually, in order not to impair crosstalk rejection.
At the same time, precision resistors ensure perfect stereo matching at maximum 0.25 dB tolerance.

Why an easily accessible balance control is important

Even high-end pickup systems may exhibit a channel level aberration of up to 3 dB.
This value represents the loudness difference between channels, resulting to a shifted stereo image - or centre position - during playback.
The fine-resolution balance control of PPA V600 allows shifting of the perceived centre position to where it actually should be, of course with a detented neutral position.
In order not to impair crosstalk rejection, balance control is achieved by fine adjustment of the right channel only. The same circuitry section is mirrored in the left channel, but without the adjustment feature. Thus, not the faintest difference between channels in terms of transit time, phase correlation, distortion and frequency response will be found.

The advantages of a variable input impedance

Considered its extremely low output voltage, the performance of a moving-coil (MC) pickup system is much influenced by the amplifier's input impedance.
The PPA V600 provides eight different impedance settings, while the ninth (47k) is the standard setting for moving-magnet (MM) systems.
The optimum setting is to be read from the pickup's specification sheet in the first place, but may also be refined by further recommendations or listening tests.
The setting is situated inside PPA V600.

The advantages of a variable input capacitance

Moving-magnet (MM) pickups with their typically high output impedance are much subject to influence by the amplifier's input capacitance.
For this purpose, the PPA V600 offers 16 individual input capacitance settings.
The optimum setting is to be read from the pickup's specification sheet in the first place, but may also be refined by further recommendations or listening tests.
The input capacitance setting is situated inside PPA V600.

What makes selectable equalization curves practical

For technical reasons, vinyl records are manufactured with an equalized frequency response, i.e. low frequencies are reduced and high frequencies are boosted.
During playback, the preamplifier must invert this equalization in order to obtain the original frequency response of the contents.
For this purpose, the equalization curve applied during record manufacture must be known.
Quite a number of equalization curves were used over the decades, however, among which the 'RIAA' curve became the most widespread - named after the Recording Industry Association of America.
Other relativly popular equalization curves one may encounter are 'NAB' (National Association of Broadcasters) or 'BBC' (British Broadcasting Corporation), which can also be properly handled by the PPA V600 when set accordingly. Both these curves have effect on treble reproduction, just like the 'FLAT' curve which does not imply high-frequency equalization at all. Furthermore, a 20 Hz high-pass or 'rumble' filter according to IEC (International Electrotechnical Commission) can be switched into circuit. See also the graph on page 17 inside the PPA V600 manual which can be found in the download area. ​​

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