REGA Tonearm modification (removing the internal ground):  REGA Tonearm modification

Which Tonearms are prepared or available for balanced operation? Tonearm-list

From engaged vinylenthusiasts is following Cartridge list:
http://mb.abovenet.de/tonbandinfo/galerie/albums/userpics/10004/TA-Daten.pdf

In the following Tonearm info list, are mounting-distance, overhang, eff. length, eff. mass and so on,
as published by the manufacturers:
http://mb.abovenet.de/tonbandinfo/galerie/albums/userpics/10004/TonArm-Liste_ohne_Bilder.pdf

NEW - correct pin alignment for phonocables using the XLR - balanced input of AQVOX Phonostages

NEW - 5-pin SME-Mini-Din ( Tiffany-Plug) Orientation / layout

Service for Audio Professionals:
Voltage divider for precise measuring of the XLR-current input of the Phono2Ci
(without this voltage divider the meter falsify/distorts the measured results)

Please contact us when you have questions or comments.

Installation and help : ASIO driver for USB, WinAmp, Foobar2000, etc..
(avoid the resampling Windows Kernel-Mixer = better audioquality from PC/Notebook
!! Apple MAC and Linux do NOT need ASIO drivers!!)

NEW - ASIO Drivers for iTunes 6 & 7 for Windows:
-- iTunes 6 - http://www.aqua-soft.org/board/showthread.php?t=32733
-- iTunes 7 - http://www.aqua-soft.org/board/showthread.php?t=38334

NEW - recording and highend music archivie building with the USB2DA
You can record all digital inputs and the microphone input of the USB2DA
with this free software. Via USB onto your computer: www.audacity.de

NEW - aditional digital inputs for the USB2DA
Some USB2DA users would like to connect more digital sources as inputs available.
We recommend to connect the best sounding device at the COAX input. Unfortunately are no quality COAX switches available.
But for the TOSlink input we have a suggestion: up to 4 devices my be connected at a single TOSlink input by using a cheap
COAX to TOSlink convertor and a automatic TOSlink switch 4in1 or 2in1.

USB-Drivers
....Problems with USB-connections often resulting from old drivers.
Here is the instruction for driver-installation:
http://focus.ti.com/lit/ug/slau238/slau238.pdf

Here are the latest drivers from TexasInstruments, the manufacturer of our USB-Interface-Chips:
USB-Codec: Texas-Instruments PCM2906 transceiver
http://www.ti.com/litv/zip/slac156
NEW - soon we complete an instruction for:
Realtime digital room correction for the USB2DA and computers
This will be very interesting... Please contact us when you have questions or comments.

A.- Read out of audio CD’s

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 http://www.exactaudiocopy.org

But also APPLES iTunes has excellent grabbing abilities, exact audiodata reading technology with low jitter.
This good sonically results are worth a try.
http://www.apple.com/itunes/ Here is the BRAKE for your CD-Rom drive. For info: most computer CD/DVD drives are reading/spinning far to fast for correct readout. This costs sound, mostly transients and details are degraded, even best read-out software can not help here.
But a brake for your drive:
For Windows:
cd-bremse http://www.cd-bremse.de (sorry, but the freeware is in german)

Für Apple MAC OS X:
DiscRotate http://discrotate.sourceforge.net/index.html



B.- Storage of audio data and data compression

First choice is the normal WAV format, but produces the biggest files, like FLAC or AIFF, nothing left out from the original.

MP3
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. Disappeared are some room- and transientinformation during compression/reduction.

FLAC and Monkey
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 therefore FLAC has far better sound quality as MP3.
Function:
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.

C.- Play back of audio data

To avoid any alterations of the audio-stream we have collected some nice PC- software tools for you:

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

Free music player software: Foobar at http://www.foobar2000.org
Configuration information for Foobar2000 at WIKI from hydrogenaudio

To achieve best possible sound under windows, ASIO drivers are required.
Installation instructions: ASIO driver for USB, WinAmp, Foobar2000, etc..

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.

With USB ASIO, DirectSound and kernel-streaming drivers you can bypass the windows K-mixer.

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

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

The most audiophile ASIO driver comes from AQVOX produces least jitter, is compatible with some other USB-DAC´s and plays with all Windows based mediaplayers/soundprograms. You can order this driver directly from us and tThe trialversion is free.

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 10x)
Ideal for grabbing purposes is the freeware programm iTunes and others.

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.

USB ASIO, DirectSound and kernel-streaming drivers
with these drivers you can go bypass 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
Please contact us when you have questions or comments.

Avoiding signal losses instead of correcting them.   The common way of reducing measurable distortions is overall negative feedback (NFB). At the first glimpse this method seems to solve the problem, but it works only for static test signals, not for "random" music signals. Negative feedback looks good on measurement equipment and paper, but has the well known limitations in sound quality we all got used to. As there are no common ways to test short time (transient) distortions, which occur during fast signal changes, most engineers assume they just don't exist. Prof. Matti Otala developed a way to measure Transient Intermodulation (TIM), which is only a small part of the whole story. It succeeded to proove the existence of short time distortions, which origin from NFB limitations. As a consequence the development of amplifiers without overall negative feedback but only local feedback and very low dynamic and static distortions was initiated. (There are already amplifiers without NFB on the market, but with sound influencing distortions.) Advanced-Class-A avoids degradation of sound quality instead of correcting it afterwards.
The breakthrough of Advanced-Class-A technology is: The signal transistor does neither pass through its voltage characteristic Vce nor the current characteristic Ic.
Passing the voltage characteristic is avoidable by using a floating cascode circuit. This method is known, but causes losses in efficiency and power when using traditional circuitry.
As the signal transistor in Advanced-Class-A circuit does not handle the current requested by the speaker, loss in efficiency is negligible.
Advanced-Class-A's most significant progress is splitting the handling of the loudspeaker's current request from the music signal voltage output stage! Advanced-Class-A means: The signal transistor is not loaded by any connected load, either devices, cartridges or speaker's current requests, because it has strong current handling "assistants" - thus no sound degradation effects due to the load happens. With today's common amplifiers it is easy to hear whether it sounds like an airy ballet dancer or a heavyweight bodybuilder, because the music signal has to pass the more or less "heavyweight" output transistors using NFB correction. In Advanced-Class-A technology the fast, airy and delicate signal transistor and the heavyweight current assistants work, supporting each other. However, the current assistants are not "allowed" to take part of the signal voltage. This means: The signal transistor does not pass through his current characteristic Ic. The load, speakers or small-signals, "sees" only the signal transistor, but no "current assistants". This is due to the signal transistor's very low output impedance, combined with the very high output impedance of the "current assistants". In case of some small current inaccuracy that may occur, the "current assistants" are safely "overruled" by the signal transistor. Unlike usual amplifiers the Advanced-Class-A amp sounds like unlimited power, as long as it works within the designated power range; combined with speed, dynamic and colourful elegance.Please contact us when you have questions or comments.

Please contact us when you have questions or comments.

Wrong cables to the turntable, or power cables too close to the phono cables/ phono stage,
can cause hum/noise or radio. It is strongly recommended to use balanced cables for the XLR-input ,
the PHONO2CI is intrinsically dead quiet.
Please play around with the position of your cables and the position of the PHONO2CI,
you then will localise disturbances that may come from other hifi devices/transformers/ handy loaders /lamps/ and so on.

You find special balanced turntable cables here
http://www.aqvox.de/cable.html#phonoinfo In case of problems please call us : +49 - 40 - 4100 6890

Please contact us when you have questions or comments.

7.- TRUE balanced with just two conductors?

Normally balanced connections have 3 conductors: positive, negative and ground.
But in the case of eg. cartriges or microphones, which are balanced sources with just 2 conductors - positive and negative. The technical term is floating balanced or balanced without ground. Yes this is TRUE balanced! The positive signal is connected to one dedicated amplifier stage and the negative signal is inverted and also connected to one dedicated amplifier stage = differential amplifier!

8.- RIAA + 50kHz Neumann Time-Constant - what is the benefit ?

technical info as pdf file

9.- DAT Bänder auslesen kopieren / SCMS , Serial Copy Management System – umgehenSome of our customers and experienced HiFi Fans or Soundstudios / Homerecorders verfügen gelegentlich noch über klanglich hervorragende DAT Bänder oder ganze DAT Archive, die sie gern auf ein anderes Medium zB. CD / DVD oder PC / Festplatte kopieren möchten. Dies ist nicht ohne weiteres möglich, da die Industrie sich damals mit dem SCMS , Serial Copy Management System, vor digitalen Audio-Kopien schützen wollte.
More info regarding this matter

Another economical possibility to bypass the SCMS comes from www.Behringer.de zB. This company produces devices like the SRC2000 Ultramatch, or Pro 24, Realtime Sample-Raten convertier, "Copy Bit Killer". The divices working as data-interfaces, means they just route the data to the AQVOX DA-converter.
CONNECT-
Simply connect the DAT-Recorder via COAX or TOSlink to the Sample-Rate converter and from there via AES/EBU oer COAX oer TOSlink into the AQVOX USB2DA DAC. From there via USB-cabel into the PC / Notebook / MAC. Thus makes it possible that you have all your music from DAT tapes on your harddive and now you can process them in your " Digital Domain".

Please contact us when you have questions or comments.

10.- Digitalization of Vinyl - RIAA to Digital - the audiophile solutioneven with the AQVOX Mic2AD the experiment to feed a Micamp or Soundcard directly with a MC/MM directly with a MC-cart will end up in frustration and in no way highend results, perhaps low-fi results.

But the Mic2AD or Micamp or Soundcard can not be blamed for this.

To apply the RIAA only with a computer, we think is not so good, because at sufficent level of bass, the mids and heighs would be overloaded by far. To equalise such a extreme unlinearity like the RIAA the basses need 40db more gain as the heighs, this is 100-times the voltage! Our Mic2AD is not able to generate 200-Volts at the output, even more not without any disturbances in signal.To leave all frequencies, except the heighs, unamplified will bring a high noise level and you are wasting 40db resolution of the AD convertor. That are from available 24bit just 17bit in theory and just 11 or 12bit in practice. That might be heard as noise.For this RIAA job, hardware is far better suitable. Our Phono2Ci eg. is ideal because of the “automatic” impedance matching function of the current amplifier topology and we find, a balanced current amplifier matches far better with a current generator (what a MC cartridge is), as a voltage amplifier ever can.The Mic2AD has balanced amplification, but is, like most Micamps or Soundcards, a voltage amplifier and there we have the impedance or capacitance problem. Means the proper termination of the MC-cartridge.For a shorter signal path you can try to connect the Phono2Ci at the RECEIVE-Inputs and select LOOP at the Mic2AD´s frontpanel. The RECEIVE input is directly connected to the A/D-converter chip.
Maybe the Phono2Ci-output/Mic2AD-AD-chip impedance matching is not as good as if you use the Mic2AD´s analog amplifier input (LINE or MIC). You need to try out.Without our Phono2Ci you will not get satisfying results. Try our Phono2Ci, it is an excellent device with some unique solutions.Sure it is no problem for a software to do the RIAA, but that is not the point. Problem is that the signal is either too noisy or too distorted. Then we have the impedance termination issue and that the AD part (like every AD-chip) in the Mic2AD does not allow any +0dbFs levels (digital overload). If you just compare the amplification value of our Phono2Ci with the Mic2AD, it looks like the Mic2AD can handle the MC amplification. But the fact that the amplification of phonostages always referres to 1kHz, the bottom needs 20db more and the top needs to be lowered by 20db which are noise now.It is not only a matter of frequency response. Frequency can be adjusted by software. And it is not a matter of software, where you have a resolution of 32Bit or even more. It's a matter of the hardware: Amplifier and A/D-converter have to handle a difference of 40dB from 20Hz to 20kHz, that means a factor of 100 in voltage. Depending on recording level, this means digital overload and / or a high noise level.40dB of frequency unbalance plus the dynamic range of the music may be to much for a recording system.In a Neumann Cutting Lathe Apparatus specially in the cutting-amplifier are lots of timekonstants, oscillators, other parts and functions which partly may be controlled by the cutter operator and which might be corrected in digital-domain.The most important filter is the 50kHz filter to save the cutter-head-coil from burn out. This timeconstant we have implemented in the RIAA of the Phono2Co. If you are interested, here is a link to this filter-part of the Neumann shematics of the Signalprocessing SAB 74B. It consists of the parts R6, R61, C35, C34 and results in 49,9kHz. This filter affects the level and the phase of signals starting at 10kHz and rising up until the end of the vinly bandwith to a maximum of 1.5db, and this is clearly audible.

11.- Upsampling and the Problem of the loudness war, digital intersample clipping +0dbFS
Es gibt ja nur wenige Hersteller von Upsampler-Chips am Markt,
zB, Analog Devices, Crystal und AKM. Keiner dieser Hersteller kann zaubern, will heissen,
wenn ein Upsamplerchip mit digital übersteuerten Musiksignalen gespeist wird, kann er technisch
nicht anders als Verzerrungen produzieren.
Das resultiert aus dem zugrundeliegenden Algorithmus, ist also kein
Qualitätsmerkmal sondern reine Mathematik.

Heutzutage werden ja leider viele Musikproduktionen nicht nur zu "tode"
komprimiert, sondern auch laut, lauter - übersteuert produziert. Übersteuert bedeutet
0dbFS+, also Digitalsignale über der eigentlich maximal möglichen Aussteuerung.
Diese Signale bringen JEDEN Upsampler zum Verzerren, hauptsächlich Phasenverschiebungen bis 90-Grad
und es hört sich logischerweise nicht audiophil an, was dann dabei herauskommt.
Abgesehen davon gehören übersteuerte Aufnahmen eher auf den Müll,
aber egal - denn den Upsampler an AQVOX DAC´s können Sie ja ausschalten.

Benachteiligt sind in dem Fall natürlich alle Besitzer von CD/DVD/SACD-Playern sowie DA-Wandlern,
bei denen sich das Upsampling nicht ausschalten lässt.Hier eine technische Erklärung von Thomas Lund, TC Electronic A/S Denmark - Stop Counting Samples.
Conventionpaper der Audio Engineering Society AS121 in San Francisco vom 08.10.06 in englisch
http://www.tcelectronic.com/media/AES121_Stop_Counting_Samples.pdfEine Erklärung in deutsch mit ähnlichem Inhalt finden Sie hier von Fritz Fey/Studio Magazin und Thomas Lund http://www.studio-magazin.de/Leseproben/Jenseits%20von%200%20dBFS.pdfHier finden Sie Informationen zu dem Thema zu laute Aufnahmen:

# Loudness War - Wikipedia
# Why Music Really Is Getting Louder - Adam Sherwin, Times Online
# How CDs Are Remastering The Art Of Noise - Tim Anderson, Guardian
Unlimited
# Brickwall Limiting - J.J. Blair, EQ Magazine
# The Big Squeeze: Mastering Engineers Debate Music's Loudness Wars - Sarah
Jones, Mix Magazine
# Everything Louder Than Everything Else - Have The Loudness Wars Reached
Their Final Battle? - Joe Gross, Austin360.com
# Why New Music Doesn't Sound As Good As It Did - Yahoo! Tech
# The Loudness War - Mark Donahue, Performer Magazine
# The Death Of Dynamic Range - Mike Richter
# Imperfect Sound Forever - Nick Southall, Stylus Magazine
# Declaring An End To The Loudness Wars - Barry Diament
# Tearing Down The Wall Of Noise - Suhas Sreedhar (Multimedia)
# Tearing Down The Wall Of Noise - Suhas Sreedhar (Text)
# Radio Ready: The Truth - Bob Orban & Frank Foti, with introduction and
comments by Bob Katz (from Katz's book "Mastering Audio: The Art And The
Science.")
# Official, Rock Music Is Too Loud - Thomas Whitaker, The Sun
# What Happened To Dynamic Range? - Bob Speer
# Pump Up The Volume - Rip Rowan, WIRED Magazine
# Over The Limit - Rip Rowan, ProRec.com
# Music Gets Louder - Adrian Larkin, BBC Radio
It's Confirmed: Music Is Really Getting Louder - Duncan Robertson, Daily
Mail
# Experts: Music Is Getting 'Too Loud' - Dave West, Digital Spy
# Distorted, Loud Rock Music Is Making Listeners 'Sick' - Adam Sherwin,
Independent News
# Masters On Mastering - JJ Jenkins, Electronic Musician
# NEW Current Trends in Recording: Is Louder Better? - Dan Banquer,
Audioholics Magazine
# NEW Loudness - Chicago Mastering Service
# NEW Whatever Happened To Dynamic Range On Compact Discs? - George Graham
# NEW Hot CD Disease - John Vestman
# NEW Music Into Noise: The Destructive Use Of Dynamic Range Compression -
Wes Lindstrom
# NEW Loudness Race Discussed - Bob Katz
# NEW How To Make Better Recordings - Integrated Metering, Monitoring, and
Leveling Practices - Bob Katz
Ein sehr interessanter Artiekl zum Thema inter sample overs, bzw. digitale Übersteuerung von unseren Korrespondenz-Kollegen der TC Electronic A/S, einem dänischen Entwickler und Hersteller von Studioelektronik 0dBFS+ Levels in Digital Mastering" by Soren H. Nielsen and Thomas Lund
http://www.tcelectronic.com/media/Level_paper_AES109.pdf
Hier grundsätzliche Infos, warum und wie 0dbFS+ beim Masteringprozess vermieden werden sollte
http://www.audioholics.com/education/audio-formats-technology/the-case-for-not-going-above-0-dbfs-for-digital-playback-systems

Aus audiophiler Sicht gehören CD´s mit übersteuerten Aufnahmen eh in die Tonne, da lässt sich auch im nachherein nichts mehr retten.
Dies betrifft leider viele aktuelle POP/Rock-Produktionen.

12.- MM oder MC - Tonabnehmer ? was ist klanglich / technisch besser?Diese Frage erreicht uns oft. MM-Systeme haben oft einen sehr starken Pegelanstieg im Frequenzbereich oberhalb 5 kHz. Je höher die Induktivität und Widerstand des MM´s je höher dieser Anstieg. Dies ist nicht nur Messbar sondern klar hörbar, zB. mehrere db Anstieg um 10 kHz, steiler Abfall bei ca. 20 kHz. wirken. Negativ treten von MM zu MM unterschiedliche Phasenverschiebungen auf, die nicht durch Phonostufen korrigiert werden können. Der krasseste Unterschied zwischen MM- und MC-Systemen ist, abgesehen von der Grundkonstruktion und Unterschiede in der bewegten Masse, das Phasenverhalten. Bei MM’s kann die Phase innerhalb des Abtastbereichs gern bis zu 180 Grad verschoben sein. Mit anderen Worten zB.: der Bassbereich wird mit korrekter Phase und der Hochtonbereich mit verdrehter Phase wiedergegeben. Das Signal hat also seine Polarität völlig verdreht.

Die Phasendrehung beträgt bei MC-Systemen jedoch stets nur wenige Grad oder Milligrad über den kompletten Abtastbereich, dort ist wohl der Grund zu suchen, warum MC-Systeme als schneller, auflösender, transparenter oder räumlicher bezeichnet werden. Technisch ist dies alles belegbar da es einfach nachzumessen ist, und natürlich nachzuhören.
Allerdings schliesst sich hier der Kreis, denn unsere Phonoverstärker sind stets mit einer erweiterten RIAA ausgestattet, die dazu beiträgt, dass ab 10kHz ebenso eine korrektere Phasenlage und Pegel erreicht wird. Man ist näher am Original, sprich hört eine natürlichere und transparentere erweiterte Hochtonwiedergabe.

RIAA + 50kHz Neumann Konstante - was bringt das ?
Hier ein pdf zum Download Zusätzliche Zeitkonstante von 3.18us bzw. 50kHz zur standard RIAA.

This question often reaches us. Mm systems have often a very strong level rise in the frequency range above 5 kHz. The more highly the inductance and resistance of the MM´s the more highly this rise. This is not only measurable but clearly audibly, e.g. several railways rise around 10 kHz, steep waste with approx. 20 kHz. work. Different phase shifts arise negatively from mm to mm, which cannot be corrected by Phonostufen. The most glaring difference between MM and MC-systems is, apart from the basic construction and differences in the moved mass, the phase behaviour. With MM's the phase can be shifted within the scanning field gladly up to 180 degrees. In other words e.g.: the bass range is shown with correct phase and the high clay/tone range with twisted phase. The signal rotated thus its polarity completely. The angular phase shift amounts to with MC-systems however always only few degrees or milli degrees over the complete scanning field, there is probably the reason to be searched, why MC-systems are called faster, more solvent, more transparently or more spatially. All of this is simple technical provably there it to check is to after-listen and naturally. However the circle closes here, because our Phonoverstärker is always equipped with an extended RIAA, which contributes to the fact that starting from 10kHz likewise a more correct phase position and level are reached. One is closer at the original, speaks hears a more natural and more transparent extended high clay/tone rendition.

13.- optimize PC´s BIOS for AUDIO-Streaming - switch off the CLOCK-spreading / clock spread option / spread spectrum - less Jitter

If you enter the PC´s BIOS you will find somewhere in the voltage/clock options the CLOCK Spreading or Clock Spread Spectrum or SPREAD Spectrum option. This function is actually meant for EMI (electromagnetic radiation) test during the CE-test of the Mainboards. If the Motherboards during the CE - certification-tests for high radiations in a certain frequency range produce to high distortions, caused by possible overlay of frequencies (also harmonic waves) and thus a reinforcement (constructional interference) of the radiated signal. This radiation behavior can be changed by letting the systemclock not longer precisely work on a specific frequency but changing its frequency very fast. The clock-frequency fluctuates and thus flattens the spikes because SPREADING it on a broader frequency band. The BIOS adjustment possibilities can look like: 0.25%; 0.5%; 1.5%; Enabled and Disabled. Disabled means the Spreading is switched off. This is the best option, however you should restart it, if radio or TV are disturbed. Disabled improves also the PC performance. Sonically effects: What the CLOCK Spreading now to do with the sound? Very simple, an instable/varying system clock produces JITTERS, all the same whether the audio data are transmitted via USB or over an internal or external sound-port or soundcard by SPdif or TOSlink.

See what wikipedia says about one of the effects of this spreading - the clock skew
http://en.wikipedia.org/wiki/Clock_skew

14.- Roomcorrection in realtime, audio streaming inside PC´s digitally corrected and played via AQVOX DAC

So called Convolving Programms or Convolvers are used for this amazing task. One good example is from Juice-HiFi the Audiolense
http://www.juicehifi.com/no/index.html

15.- Audiosoftware AQVOX recommends for recording, cutting, mastering.. 96, 192 kHz 16 / 24 / 32 / 64 bit
www.n-track.com is a fair priced but professional multitrack Software for recording , arranging, mixing and so on...
Works with VST-plugins, Direct-X, Re-Wire, ASIO and deals with lots of file formats up to 192kHz,

www.audacity.de - is a freeware Audioeditor and Recordingsoftware for creating, recording, manipulating, editing and cutting of a lot of different file formats. Good fro the unexperienced user.www.steinberg.net - wavelab is the standard for audio editing.