La verita' sull'hifi:quello che .....

Pre Phono con pilotaggio in corrente per testine Moving Coil


Via rca input ci sono le ampie regolazioni che si vedono sul pannello posteriore,per pilotaggio in tensione di testine MagnetoMobile..

Via XLR,usando l'adattatore rca xlr optional,per le testine MC ha particolarmente senso,la regolazione e' automatica e i dip switch non servono,in tal caso il pilotaggio e' in corrente.

Unico pre phono a pilotaggio in corrente a cifre accettabili.

By soundstage:thank you.

Aqvox Phono 2 CI Phono Stage



Review Summary

Sound

"A neutral, even-handed performer with surprisingly potent bass and astounding retrieval of detail considering its modest price." "From slamming rock to delicate jazz, from art-noise to full-throttle classical, the Aqvox Phono 2 CI handled it all with aplomb." "The Phono 2 CI rose to the challenge, rendering itself essentially invisible in a distinctly non-budget manner."

Features

"Circuit-wise, the Phono 2 CI runs in pure class A and uses no op-amps in the signal path." "The front panel contains the power switch; a well-chosen rumble filter (it kicks in at 10Hz); left and right gain knobs, which are much more logical and easier to deal with than DIP switches (although there are DIP switches around back for gross adjustments, with up to 68dB of gain available)." Around back "there are both single-ended and balanced inputs and outputs" and "a comprehensive set of DIP switches for both capacitance and impedance."

Use

"While the Phono 2 CI does have single-ended connections, it is fully balanced from one end to the other, and Aqvox strongly suggests that these connections be used in order to allow the Phono 2 CI to perform at its best." "I can't imagine that there would be a cartridge out there that would give the Phono 2 CI much trouble in the gain department."

Value

"If you're in the market for a sub-$1000 phono stage -- heck, even if you have twice that to spend -- you have to check out the Aqvox Phono 2 CI."


The arrival of a component from a hitherto unknown company is an event fraught with excitement and peril. Excitement? Who doesn't like a surprise? Decanting an electronic product is fun enough, but add in the spice of the unknown and you've got the recipe for a racing pulse.
But peril? What kind of peril could possibly accompany a piece of audio gear? Well, maybe peril is a bit strong, but with products from unknown companies, we reviewers have no point of reference, so the work of understanding such components is harder. Personally, I flounder around for awhile, trying to determine, in a vacuum, where the product fits in. Also, if it seems to work either very well or very poorly, I often end up second-guessing myself, wasting valuable reviewing time trying to figure out if this thing is the audio equivalent of the Rosetta Stone or just a James Ossuary.
Well, let me tell you about the Aqvox Phono 2 CI phono stage from Germany, which is passing along analog signals from across the years in my listening room right now. Never heard of it? You will.
The details
Aqvox is a relatively new German company that's pretty much unknown in audiophile circles. Suzanne Candeias founded the company in 2003 with the intention of offering inexpensive recording and reproduction components to studios and multimedia companies. But, as we shall see, Aqvox products have some serious crossover appeal. In addition to the Phono 2 CI, Aqvox also produces a USB-connectable DAC and a microphone preamp with an A/D converter.
The most immediately distinctive feature of the $900 USD Aqvox Phono 2 CI is its firm reliance on both balanced outputs and inputs. While the Phono 2 CI does have single-ended connections, it is fully balanced from one end to the other, and Aqvox strongly suggests that these connections be used in order to allow the Phono 2 CI to perform at its best. Phono signals are naturally balanced, with a positive signal, negative signal and ground, and although most phono stages simply shunt that negative signal off to ground and use only the positive, there's some serious merit to the idea that the miniscule signals that squeak out of a moving-coil cartridge could use all the noise-canceling help they can get. While it's not that hard to reterminate your RCA connections with XLR plugs, Aqvox thoughtfully includes well-made RCA-to-XLR adapters along with the Phono 2 CI so you can get going right away. I tried them, and they seem to work quite well.
Considering its price, the Phono 2 CI is reasonably well made, with a CNC-milled aluminum front panel, extruded-aluminum side panels, and all-aluminum chassis work. At a touch over 6 pounds, the Phono 2 CI is not heavy, but it does feel reasonably solid. The front panel contains the power switch; a well-chosen rumble filter (it kicks in at 10Hz); left and right gain knobs, which are much more logical and easier to deal with than DIP switches (although there are DIP switches around back for gross adjustments, with up to 68dB of gain available).
'Round back, things get somewhat more intricate. There are both single-ended and balanced inputs and outputs, although you can only use only one set of inputs at any one time. Power arrives courtesy of an IEC connector. There is also a comprehensive set of DIP switches for both capacitance and impedance, but these aren't as straightforward as they may seem.

By AQVOX,precisazioni sull'uso.

your MC will work well.
If you go in balanced, you do not need to adjust any switch at the packpanel -
except activate the XLR-input switch. That is all, because the impedance will be matched automatically.
But if you go in XLR, you can still adjust the impedance and capacity manually, by activating the RCA-input!!
The only thing you need to adjust you need to find the best sound with the frontpanel INPUT-GAIN-KNOBS. These frontpanel knobs are only for matching the input with the cartridge. The frontpanel knobs are NOT nolumeknobs. if you use them wrong, you get bad sound.
Best position is where singer voices coming out of the speakers and focussing in front of you.
This frontpanel INPUT-GAIN knobs are for adjusting both inputs (XLR or RCA) and for MM and MC.
The XLR input is only for MC (this is the unique balanced current amplifier)
The RCA inout is for MC and MM (this is the unbalanced voltage amplifier)
You can only connect one cartridge at one time. Do not use both inputs at the same time.
You can use both outputs at the same time.

BY SIXMOONS.com:

The current 950 euros charged (the upgrade from MkI to II is 300 euros) remain quite competitive for a fully balanced machine. Which naturally begs the question - is balanced really required? Aqvox responds with four bullets:

1. Symmetry
Because cartridges are inherently balanced, this advantage should be - um, taken advantage of. For one, it means reduced signal transfer distortion, sensible with the miniature voltages involved. Then there's the +6dB gain advantage since balanced amplification doesn't reference to 0 but anti phase.


2. Current gain
According to Aqvox, MC carts pass on "decent current" but "negligible voltage". There's little current change between high- and low-output MCs unlike voltage. Hence current gain becomes choice, which for MC picks up two advantages: the pickup is electrically damped (analogous to damping factor) which helps MC's bothersome HF resonance without undermining dynamics. Secondly, impedance matching for MCs become redundant as long as those (and only those) are run purely symmetrical. Plug 'n' play is the term, putting an end to audiophile night sweats and panic over "damn, perhaps another 20 ohm". Rather, our audiophile will grumblingly pull the comforter over his ears because you stole one of his toys...


3. RIAA plus Neumann
While the RIAA EQ curve is the official standard, plus Neumann is the unwritten law. Mister Neumann at the time limited
HF boost to protect the cutter heads of his cutting lathe. Having become the facto standard, this should be likewise employed on the re-EQ end, less attenuation above 10kHz than pure RIAA would have it. A rational argument.


4. LEF
This pertains to the Candeias gain modules also seen with C.E.C. To combat transistor nonlinearities and concomitant nonlinear gain, Load Effect Free circuitry (a form of floating cascade circuit) avoids running the voltage and current load lines through the output transistors. This is claimed to linearize gain without global feedback while cleaning up the sound. Further details on this can be found here. All Aqvox components (i.e. also the D/A converter and microphone preamp) rely on LEF. The Phono 2Ci sports four Candeias ICs, one per channel and gain block. Any reason for the disco lighting above though?




What else? Three buttons and two rotary knobs. MM/MC selection is on the back panel as is ground lift to avoid hum. The subsonic filter is upfront as are the dual mono gain trim pots. "When dimensionality and soundstaging lock in, you've got the perfect gain value" assures Aqvox's Herr Lübke, differentiating the importance of this setting from your preamp's or integrated's master volume which controls ultimate playback levels.


Needless to say, symmetrical circuitry requires symmetrical cabling. That's standard with most tone arms (Rega excepted - refer to the Aqvox home page for conversion). Should your interconnects terminate in RCA -- standard -- 25 euros add an RCA-to-XLR adapter. That's what I used to equalize things for comparative purposes. It also avoided readjusting VTA endlessly. My phono cable exits directly beneath the tonearm to require dismounting to change cables. True balanced operation with its attendant noise reduction is audible in practice, however. Backgrounds get blacker, hum drops and I'm not really missing the Russian radio news announcer during album swaps.

The Aqvox Phono 2Ci MkII meanwhile offers:
Great stage width and depth with accurate localizations but no undue chiseling.
The treble is in its own class: clear, beyond criticism, soft without indecision, perfectly balanced.
Ditto for the mids: Vocal balance is spot on, neither too close nor too far, very very accurate without spittle hitting the microphone - unless it really did. Highly nuanced.
Only the bass wants for the occasional dose of pepper not for quality -- grip, articulation and timing are spot on -- but simply raw quantity.
Already the most detailed, nuanced and graduated, it's also microdynamically the most accurate of the three. The Aqvox is highly informative, weaving a dense net of apparently secondary sonic trivia yet exactly this -- plus brilliant staging -- adds up to the credible 'live effect' I appreciate. Macrodynamics are good but the SAC is better still.
For the money, fit and trim are first rate, the latter probably due to the pro roots - unpretentious and solid, with balanced and single-ended i/o ports, ground lift, variable gain in front, around back and dual mono. Add four selectable capacitance values, only three for input impedance (because MC should use XLRs).
On the latter point, if you own more MCs than inputs, the Aqvox 'auto-matching' feature is key (the RCA input, to my ears, is indeed a few degrees more subdued).

In a perfect world, the Aqvox would be fitted with a bass boost button. I wouldn't use it a lot but occasionally. If it were called gamma rather than bass boost, I'd simply grin mischievously.

AQVOX dac usb

Questo e' un convertitore DAC da sogno!I suoi 950 euro sono ben spesi!
Dettaglio e dolcezza ai massimi livelli.
Testato con cd Monrio+AQVOX DAC+Pre Quad+Finale Aeron 8900+ Audes Excellence 5.Sto ascoltando senza oversampling.


The USB 2 D/A is one of the best USB DAC´s available with selectable an upsampling to 24/192 DAC - Digital to Analog Converter for use in high-end or pro audio environment.
Supports up to 24bit/192kHz sampling frequency via S/PDIF COAX and AES/EBU, TOSLink and USB. Furthermore it is designed to work as an external computer soundcard via USB 1.1 connection (no drivers needed for windows or apple).
With the integrated audiophile headphone amplifier all digital inputs (even USB) can be monitored.
The USB2DA improves the audio signal of any CD/DVD Player, PC, Notebook, DAW´s, WiFi Player etc...all connectable digital audio sources.

Compatibility:
The USB input/output is compatible with Windows98SE, Windows 2000, XP, Vista, Linux, Mac OS

4 Digital inputs: AES/EBU, COAX, TOSLINK and USB (USB bridges up to 10m). Für längere Strecken benötigt man USB-Repeater bzw. USB Booster, die sich hintereinander schalten lassen.)

2 Analog outputs: balanced XLR and unbalanced RCA.

1 Digital output: COAX, TOS, AES/EBU-Input are routed to the USB-output (up to 48kHz/16bit).

USB Formatconverter
Converts from and to following formats : USB in/out maximum 48kHz/16bit
USB to SPDIF - S/PDIF
SPDIF - S/PDIF to USB
COAX to USB
USB to COAX
TOSlink to USB
USB to TOSlink
AES/EBU to USB
USB to AES/EBU
USB to Analog symmetrical / balanced
USB to Analog unsymmetrical / unbalanced
Analog symmetrical / balanced to USB
Analog unsymmetrical / unbalanced to USB
PCM to USB
USB to PCM
digitalsymmetrical / digital balanced to USB
digital out to USB
Cinch to USB
USB to Cinch

No Op-Amps in signal path
The AQVOX USB 2 D/A was designed using innovative discrete technology. Only single transistors are used in the amplification stages. No integrated circuit amplifiers (OP-amps) and the latest state of the art components to sound most dynamic, highly detailed yet neutral and untiring. We believe the USB 2 D/A to be a unit with outstanding sound quality and an excellent value.

NEW in the MKII Version:
Headphone amplifier in audiophile quality and USB headphone amp.
The fully discrete Class-A circuitry has no IC in signalpath and
delivers exceptional soundquality at actual high end niveau.
A first class monitoring solution. The headphoneamp is coupled directly to the DAC chip.
This saves the voltage gain stage - a shorter signalpath is almost not possible.

No global overall feedback
Fully discrete (single transistors) Single Ended Class A amplification technology and the use of just one amplifier per balanced analog output stage provides you with today's best possible sound quality. The amplifier section runs without global overall negative feedback, only lokal FB is used. Thus the amplifier generates drastically less dynamic distortions.

JitterEx precision-reclocking 192kHz/24Bit upsampling
Before the digital to analog conversion process, all signals are upsampled and converted to 192kHz/24Bit (unless the bypass is activated). This procedure is advantageous because DACs can offer better performance at higher sampling rates. Upsampling enables filtering to take place far beyond the range of human hearing, as well as offering other, audible benefits, right across the audio band, including improved transient response and less ringing. This results in a more open, transparent sound, tighter bass and a generally more "musical" sound.
NEW in the MKII Version: by factor 15 improved Clock.

True balanced circuitry for entire analog signal path
Featured with an AES/EBU digital input and a balanced XLR analog output the USB 2 D/A is the best choice for professional users to monitor digital recordings. True balanced circuitry for entire analog signal path. The USB 2 D/A is also excellent for persons who strive for the highest level of sound quality from their home audio systems, as well as for computer users who like to get the best output quality from their Workstations or Notebooks. No need to use a low-fi inbuilt soundcard. Furthermore users can upgrade their micro hifi system (portable CD, MD... as far as they provide a digital output) to a real high-end solution.

NEW in the MKII Version: LowNoise power supply, with 10dB quiter power output.
Our designers created a completely new LowNoise powersupply. Now the power output delivers with 10dB lower noise, cleaner energy and has a higher efficiency. Some better prefilters where added to bann the dirt also in lower frequencys. At the powersupplys output resides a linear regulator. The result is audible. And the sound of the MKII is now better even under bad power conditions.



To achieve best possible sound under windows, ASIO drivers are required.
VISTA, Apple MAC and Linux do NOT need any ASIO drivers!

Installation instructions: ASIO driver for USB, WinAmp, Foobar2000, etc..

USB ASIO, DirectSound and kernel-streaming drivers
with these drivers you can go around the windows K-mixer.


The K-Mixer (kernel-mixer) is the mixing console inside windows, where all sound "channels" of a PC are routed trough.
Unfortuntely the K-mixer has an ( for audiophile and pro-users ) unwanted and audible habit, he resamples the audio signal.
Apple computers with USB connection do not have this k-mixer.

We have collected some nice PC-audio software tools for you:

Audio-files playback software with flexible outputs and free driver support


Free music player software: WINAMP5 : http://www.winamp.com


Free music player software: Foobar :http://www.foobar2000.org
configuration information for Foobar2000 at WIKI from hydrogenaudio
http://wiki.hydrogenaudio.org/index.php?title=Foobar2000


Freeware ASIO driver:
http://www3.cypress.ne.jp/otachan/
(unpack files in .7z format with 7-ZIP)

Freeware ASIO driver:
http://www.asio4all.de

This USB ASIO driver is sold by the company USB-AUDIO, and can be bought online for ca. 50.- Euro:
http://www.usb-audio.com

Here is the BRAKE for your CD-Rom drive. Most PC CD-Rom drives are making to much noise when playing audio CD´s,
because they do not speed automatically down.
www.cdbremse.de (sorry, but the freeware is in german)

Computers, PC´s, Notebooks and other devices with an USB interface may be used as "audio-datasources ":
(the device must provide a so called"USB-Host" feature) Some devies need an firmware update to provide the required functionality, please check with the manufacturer or dealer.

About the matter: read out of Audio CD´s in highest quality and store or playback from a PC or MAC harddrive.

How and with what should be grabbed? ( Audio CD read out)
The (cleaned) Audio CD´s should be grabbed with slowest possible readout speed.
(for info: some CD/DVD devices in computers are not able to read out data with speeds under 4x)
Ideal for grabbing purposes is the freeware programm Exact Audio Copy (EAC) http://www.exactaudiocopy.de
Excellent error correction and bit-perfect: http://www.exactaudiocopy.org

EAC incorporates the most exact audiodata reading technology with the least jitter of all audiograbbing programms available now.

How and with what should be compressed?:( audio aata loss-less compression )
First choice is the normal WAV format, which produces the biggest files of all compression programms.
Here is nothing left out from the orioginal.
MP3 is NOT suitable for highest soundreproduction and highend applications. Even at lowest compression rates where just 10% of storing space (compared to the original WAV file) is saved, results in smaller sound stages with less depth. Here are room- and transientinformations left out during compression/reduction losses.

The compression programm FLAC ( http://flac.sourceforge.net ) and also Monkey belong to the group of Loss Less (only compression no reduction) compressors and saving ca. 50% storage space. This is far less as with a MP3 reduction can be saved, but without any losses, because a FLAC file can be restored to 100% of the original WAV file, and FLAC has far better sound quality as MP3.

The ideal case looks like: CD is in the CD-drive, EAC finds the CD title in the CD database and writes all titles automatically and reads out the CD. FLAC is integrated in EAC and EAC writes the title-tags in the FLAC files. It can´t be much easyer and convinient!
The advantage of loss-less compression is, that the audiosignal is after the decompression 100% identical to the original music signal.

Fact list

1. No Op Amp in signal path

2. No global Feedback-loop

3. Dual Mono DAC (2 x Burr Brown PCM1796)
results in endless (optimal) channel-separation, more dynamic jet stability and less than neglible noise.

4. Current Amplification DAC to analog output-stage coupling.
reduced amplification-stages, and driven in pure class-A operation.

5. Only passive elements are used in the filter-stages. Advantage: TOP-grade measurement-figures.
This technic/design is unique.

6. Passive analog filters
produces -because on principle- no dynamic distortions.

7. Single ended Class A amplifiers
are known for musical and dynamic sound. But require Class A layout.

8. Sample rate converted to 195kHz independent of input sampling.
independent of input frequency, ultra stable 195kHz clock produces extremely low jitter signal.

9. Precision internal clocks for jitter elimination.
Temperature and ageing compensated oscillator-modules.

10. 4 digital inputs: AES/EBU, RCA (16bit/32kHz till 24bit/192kHz), Toslink (16bit/32kHz till 24bit/96kHz), USB 1.1 (6kHz till 16bit/48kHz) USB1.1 (USB2.0 compatible) Plug-n-Play no drivers needed (WINXP, WIN2000, WINSE, WIN98, and Mac OS X).

11. Independent integrated PC USB 1.1 sound system functionality. The USB 2 D/A can be used also like a external soundcard.
Via the USB-connection all digital inputs and the analog Microphone input can be ein tentransfered into the computer
Here information how to make digital recordings from the USB2DA to a computer.

12. Digital data transfer from Microphone input to computer via the USB port. With a measurement microphone and software like
CARA ( http://www.cara.de/ENU/index.html ) the analysis / optimizing of the room acoustics or speakers is possible.

13. A/D converter for USB with Microphone amplifier and 12-volt phantom power.

14. all digital inputs are routed to the USB output. (COAX / Toslink / AES/EBU to USB)

15. XLR and RCA outputs.
True balanced circuitry for entire analog signal path.

16. fully discrete (no OPamp) headphone amplifier in Class-A

17. Sampling frequency indicator.

18. Digital de-emphasis.

19. Flat and dynamic roll off digital filter.

20. Selectable dither.

21. 3 selectable oversampling ratios.

22. 100V ...240V operating AC voltages.

23. Blue LED backlighted push-buttons.

24. Ultra clean switching power supply
with special pre- and post filtering of AC power voltage.

25. Insensitive to AC voltage distortions, DC fragments, noise and voltage stability.

26. Full aluminium casted turning knobs with long-distance viewable pointers.

27. Full Aluminium extrusion casted and machined front panel and side profiles.

28. All capacitors in signal path are finest inductive free-type very low loss
polypropylene capacitors.

29. Biological friendly lead-free CuAgSn silver solder.

30. PCB boards completely gold plated.

31. 1% metal-film resistors are used within the audio path.

32. 19" full-aluminium robust rack-mounts are available.

33. External dimensions (w / h / d): approx. 435 x 59 x 290 mm

34. Weight approx. 2.8 kg