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Things to know ...

 ... about Headphone Amps


... about Phono Pre-Amps

 ... about D/A Converters

... about Headphone Amps

Which advantages do balanced 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 (180 degrees phase 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.
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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.
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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.
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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.
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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.
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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.
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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 HPA 100 and HPA V200.
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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.
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Why is a 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.
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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.
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Why makes it 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.
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About the digital input options from V100 / V181 / V200

The headphone amps V100 / V181 / V200 may be equipped each with one of three ditial inputs.
OPTO 24/96 will accept data with 24 bit and 28 ... 110 kHz sample rates .
USB 24/96 will accept data with 24 bit and 44.1, 48 and 96 kHz sample rate.
USB 16/48 will accept data with 16 bit and 44.1 and 48 kHz sample rate.

As no input selector is provided, the switching over to the digital signal is done automatically when a signal is present.
This means that a digital signal has priority over any analog signal connected to the headphone amp.

If a headphone amp is ordered with the digital input option, this will be installed before delivery.
But also retrofitting a digital input option is possible and relatively simple after purchase.
If in doubt please order the installation manual in advance by mail !


About the optional OPTO input.

The OPTO 24/96 option will accept data with 24 bit and sample rates streching from 28 ... 110 kHz. The TOS-Link optical receiver forwards the data to a digital receiver circuitry and from here to a D/A converter offering 110dB dynamic range and 100dB THD+N. With these parameters, the achievable quality is much better than the 16-bit CD standard and is nearly adequate to the high overall analog performance of the headphone amps.
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About USB

Which USB devices can be connected ?

Since the the optional USB inputs for V100 / V181 / V200 are terminal devices after USB regulations, they have a type-B interface. Connections can only be established to hosts (tabletop or laptop PCs). MP3 players or similar gear cannot be connected digitally via USB. Of coarse you may connect these devices in the analog domain.
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What about the quality of the USB input ?

The USB 16/48 option is operated at 16 bit and (usually) 44.1kHz sample rate. The USB receiver forwards the data to an integrated D/A converter offering 105dB dynamic range and 95dB THD+N. With these parameters, the achievable quality conforms the 16-bit CD standard, but is somewhat worse than the overall analog performance of the headphone amps.
The USB 24/96 option will accept data with 24 bit and 44.1, 48 and 96 kHz sample rate. The USB receiver forwards the data to a separate D/A converter offering 110dB dynamic range and 100dB THD+N. With these parameters, the achievable quality is much better than the 16-bit CD standard and is nearly adequate to the high overall analog performance of the headphone amps.
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Why does 100% host volume setting make sense ?

Volume control within the host is always accomplished digitally, i.e. bits are removed from the data stream. Therefore a signal attenuated by 24 dB will cost 4 bit of performance. When only 16 bit signals are present this would end up in sound issues !
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Can the digital signal be utilized any further ?

When the unbalanced cinch sockets of the dedicated headphone amp is configured as outputs, the converted USB signal is available there in analog form and can be used to feed further external audio devices, like amplifiers e.g.
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About headphone amps with balanced outputs
Regarding common amplifiers with phone jacks (with unbalanced outputs) the remaining energy from the headphones will find its way through a shared ground cable back to the internal ground which both have specific resistances. Whilst doing so, the ground - which should remain calm - is "polluted" by remains of left and right signals.
This means crosstalk between both channels and it is hearable and measurable !
In balanced mode the voice coil from the headphone is driven by two amplifiers. One is operating the normal signal, one the 180 degree phase shifted (inverted) signal what results in a "push-pull" operation. Whilst one amp is "pushing" the voice coil, the other amp is "pulling" it. Doing so, a double as high output voltage is accomplished, creating a much higher volume. As the wanted additional effect now the ground is free of any influences as the energy flows only between both operation voltages.
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... about Phono Pre-Amps

Why are turntables 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.
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What's the advantage of a balanced signal ?

In contrast to unbalanced signals, balanced signals are transmitted via two wires plus ground connection. A balanced signal is accomplished by inverting the original source signal at unaltered level. One wire carries the original, unaltered signal (a), the second wire the inverted signal (-a).
At the receiving input, the signal is fed to a differential amplifier which 'sees' the difference between a and -a only, at a resulting level of 2*a.
On a signal path between devices, interferences (s) may try to disturb the signal by entering the connecting line, but being in-phase with respect to both signal wires. The differential input again sees the difference between interferences on both wires only. Thus - after the rule (s) - (s) = 0 - they are eliminated almost completely.
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Why did we choose instrumentation amplifiers ?

Like mentioned earlier, the balanced signals are fed to special instrumentation amplifiers with differential inputs, which were originally intended for the use with microphones. 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.
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Why is an easily accessible gain control 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. Adjustment is provided for both channels individually, in order not to impair crosstalk rejection. At the same time, precision resistors ensure perfect stereo matching at 0.25 dB tolerance.
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Why is easily accessible balance control 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 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.
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What are the advantages of 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.
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What are the advantages of variable input capacity ?

Moving-magnet (MM) pickups with their typically high impedance are much subject to influence by the amplifier's input capacity. 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.
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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 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.
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Why are individual signal paths so favourable ?

Most operational amplifiers are designed as two amplifiers integrated in a single package. If the left and right channel signal are processed by the circuit at the same time, a certain mutual interference is inevitable. The adverse effects may be marginal only, but should be excluded during the design process wherever feasible.
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Why are operational amplifiers ideal for small-signal processing ?

High-end audio gear often follows a 'discrete' design, meaning that the amplifier stages are composed of single transistors. This feature is often presented as an optimizing measure, but finally the related high manufacturing effort is to be paid by the customer.
Certainly an operational amplifier is made of transistors as well. Its design however offers the advantage of thermal coupling of its individual components, and ageing issues play a negligible role.
Since the choice of OP-amps on the market is huge, an ideal model for any special application can be found.
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Why are switching-mode power supplies ideal for phono preamps ?

An equalizing preamp boosts low frequencies and therefore does right the same with hum interference radiated by a mains transformer. Thus, conventional transformers are 'audible' in many cases, sadly...
One possible solution would be the use of a remote power supply, which of course means additional costs for the customer due to an extra device and cabling.
As an equalizer preamp consumes little power at quite constant current, and short ways between current source and load generally make sense in terms of voltage decoupling, a switching-mode power supply is the most intelligent solution here.
The PPA V600 contains two thereof, i.e. one for each positive and negative voltage supply. They both provide 24 volts, which are then reduced to well-backed +/- 18 volts by linear regulators. Switching frequency ranges well beyond 100kHz, and highly sophisticated filtering between the raw-voltage supply and linear regulation avoids any switching-frequency leakage into the signal section.
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... about D/A Converters

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.
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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 – much more than can be found in real life !!
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.
The best today AD converters offer dynamic ranges from 120 dB with distortion figures about –110 dB THD.
Lots of losses have to be faced during recording, editing, mixing …   
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.
When a 16 bit CD signal is input to a 24 bit DA converter, this signal may be attenuated by 6 dB x 8 Bit = 48 dB = factor 200:1 WITHOUT changing anything from the original data.
We learned from the above 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.
And– for our opinion - digital attenuation is the best what can happen to a signal (except not being attenuated).
Of coarse 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.  
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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.
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