WEISS MEDEA User manual

Weiss Engineering Ltd.
Florastrasse 42, 8610 Uster, Switzerland
www.weiss-highend.co
MEDEA+
OWNERS MANUAL

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INTRODUCTION
D ear C usto er
Congratulations on your purchase of the MEDEA+ D/A Converter and welco e to the fa ily of
Weiss equip ent owners!
The MEDEA+ is the result of an intensive research and develop ent process. Research was
conducted both in analog and digital circuit design, as well as in signal processing algorith
specification.
On the following pages I will introduce you to our views on high quality audio processing. These
include funda ental digital and analog audio concepts and the MEDEA+ converter.
I wish you a long-lasting relationship with your MEDEA+.
Yours sincerely,
Daniel Weiss
President, Weiss Engineering Ltd.

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TABLE OF CONTENTS
4 A short history of Weiss Engineering
5 Our ission and product philosophy
6 Advanced digital and analog audio concepts explained
6 Jitter Suppression, Clocking
8 Upsa pling, Oversa pling and Sa pling Rate Conversion in General
11 Reconstruction Filters
11 Analog Output Stages
12 Dithering
14 The MEDEA+ D/A Converter
14 Features
17 Operation
20 Technical Data
23 Contact
24 Addendu for the FireWire option
32 Addendu for the ST Glass Fiber input option

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A SHORT HISTORY OF WEISS EN INEERIN
After studying electrical engineering, Daniel Weiss joined the Willi Studer (Studer - Revox)
co pany in Switzerland. His work included the design of a sa pling frequency converter and of
digital signal processing electronics for digital audio recorders.
In 1985, Mr. Weiss founded the co pany Weiss Engineering Ltd. Fro the outset the co pany
concentrated on the design and anufacture of digital audio equip ent for astering studios. Its
first product was the odular "102 Series" syste . After 23 years, this syste is still up to date
(24 bit / 96kHz) and is still being sold. Hundreds of Mastering Studios around the world use it
every day.
In the early nineties the „Ga bit Series“ was launched, taking ergono ics and sonic quality to new
heights. The „Ga bit Series“consists of stand-alone units like Equalizer, Denoiser / Declicker,
Dyna ics Processor, A/D converter, D/A converter, Sa pling Frequency Converter, Dithering etc.
40 bit floating point processors and sa pling rates up to 96kHz are e ployed.
In 2001 we have decided to enter the High-End Hi-Fi arket which offers a co parable clientele
to that of the Mastering Studios. Both consist of critical and discerning listeners, who are in
constant search for the best audio reproduction equip ent or the best audio tools respectively.
Our list of clients includes big na es, like SONY, BMG, EMI, Warner, Hit Factory, Abbey Road,
Teldec, Telarc, Unitel, Gateway Mastering (Bob Ludwig), Bernie Grund an Mastering, Masterdisk,
Sterling Sound, Whitfield Street, Metropolis and hundreds ore.
For a ore co prehensive list you are invited to visit our pro audio website at www.weiss.ch.

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OUR MISSION AND PRODUCT PHILOSOPHY
The wealth of experience we have gained in over 20 years of designing products for top Mastering
Engineers, we now apply to the design of outstanding High-End Hi-Fi products.
Our ission is to create equip ent which beco es classic right fro the outset; - outstanding in
sonics and design.
These are some of the milestones at Weiss Engineering:
1985 Introduction of the "102 Series", a 24 bit odular digital audio processor for Mastering
Studios
1986 Introduction of one of the first sa ple rate converters for digital audio
1987 Introduction of one of the first digital equalizers
1989 Introduction of one of the first digital dyna ics processors
1991 Introduction of the "Ibis" digital ixing console, built for the ix-down of classical usic
1993 Introduction of the "Ga bit" Series of digital audio processors, which e ploy 40 bit floating
point processing and sport an extre ely ergono ic user interface
1995 First 96kHz sa pling rate capable products delivered
2001 Introduction of the MEDEA, our High-End Hi-Fi D/A converter and the first product in our
High-End Series
2004 Introduction of the JASON CD Transport
2007 Introduction of the CASTOR, our High-End Hi-Fi Power A plifier
2008 Introduction of the MINERVA Firewire DAC, VESTA Firewire Interface, ATT202 passive
attenuator
2010 Introduction of the DAC202 DAC, INT202 Firewire Interface, MEDEA+ DAC

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ADVANCED DI ITAL AND ANALO AUDIO CONCEPTS EXPLAINED
Jitter Suppression and Clocking
What is jitter and how does it affect audio quality? In the audio field the ter jitter designates a
ti ing uncertainty of digital clock signals. E.g. in an Analog to Digital Converter (A/D) the analog
signal is sa pled ( easured) at regular ti e intervals; in the case of a CD, 44100 ti es a second
or every 22.675737.. icroseconds.
If these ti e intervals are not strictly constant then one talks of a jittery conversion clock. In
practice it is of course not possible to generate exactly the sa e ti e interval between each and
every sa ple. After all, even digital signals are analog in their properties and thus are influenced
by noise, crosstalk, power supply fluctuations, te perature etc.
Hence a jittery clock introduces errors to the easure ents taken by the A/D, resulting fro
easure ents being taken at the wrong ti e. One can easily observe that the level of the error
introduced is higher during high audio frequencies, because high frequency signals have a steeper
signal for .
A good designer takes care that the jitter a ount in his/her design is ini ized as well as
possible.
What type of equip ent can be co pro ised by jitter?
There are three types: The A/D Converter as described above, then there is the D/A Converter
where the sa e echanis as in the A/D Converter applies and the third is the Asynchronous
Sa ple Rate Converter (ASRC). The ASRC is not so ething usually found in Hi-Fi syste s. It is
used by Sound Engineers to change the sa ple rate fro e.g. 96kHz to 44.1kHz, or e.g. for
putting a 96kHz recording onto a 44.1kHz CD.
You ay now argue that in High-End Hi-Fi there are such things as „Oversa plers“ or
„Upsa plers“.

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Yes, those are in essence sa pling rate converters, however in a well designed syste these
converters e ploy a synchronous design, where jitter does not play any role. Of course a
conversion between 96kHz and 44.1kHz as in the exa ple above, can be done in a synchronous
anner as well. An ASRC in fact is only required either where one or both of the sa pling
frequencies involved are changing over ti e („varispeed“ ode of digital audio recorders) or where
it is unpractical to synchronize the two sa pling frequencies.
So basically in Hi-Fi jitter atters where there are A/D or D/A converters involved. CD and DVD
players are by far the ost nu erous type of equip ent e ploying D/A converters. And of course
stand-alone D/A converters. Jitter, being an analog quantity, can creep in at various places. The
D/A converter built into CD or DVD players can be „infected“ by jitter through various crosstalk
echanis s, like power supply conta ination by power hungry otors (spindle / servo) or
icrophony of the crystal generating the sa pling clock or capacitive / inductive crosstalk between
clock signals etc.
In the standalone D/A converter jitter can be introduced by inferior cables between the source
(e.g. CD player) and the D/A converter unit or by the sa e echanis s as described above except
for the otors of course.
In the case of a stand-alone D/A converter (as the MEDEA+), one has to take two different jitter
conta ination pathes into account.
One is the internal path where internal signals can affect the jitter a ount of the sa pling clock
generator. Here, all the good old analog design principles have to be applied. Such as shielding
fro electric or agnetic fields, good grounding, good power supply decoupling, good signal
trans ission between the clock generator and the actual D/A chip.
The other path is the external signal co ing fro the source to which the sa pling clock has to be
locked. I.e. the D/A converter has to run synchronous to the inco ing digital audio signal and thus
the frequency of the internal sa pling clock generator has to be controlled so that it runs at the
sa e sa pling speed as the source (CD transport). This controlling is done by a Phase Locked
Loop (PLL) which is a control syste with error feedback. Of course the PLL has to be able to follow
the long ter fluctuations of the source, e.g. the sa pling rate of the source will alter slightly over
ti e or over te perature, it will not be a constant 44.1kHz in the case of a CD. But the PLL should
not follow the short ter fluctuations (jitter). Think of the PLL as beeing like a very slow-reacting
fly-wheel.

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In the MEDEA+ we e ploy a two stage PLL circuitry which very effectively suppresses jitter. A
co on proble with ost PLLs used in audio circuitry is that they suppress jitter only for higher
frequencies. Jitter frequencies which are low (e.g. below 1kHz or so) are often only arginally
suppressed. It has been shown that low frequency jitter can have a large influence on the audio
quality though. The MEDEA+ suppresses even very low frequency jitter co ponents down to the
sub-Hertz range.
This eans that the MEDEA+ is virtually i une to the quality of the audio source regarding jitter.
For a CD player as a source this eans that as long as the data is read off the CD in a correct
anner (i.e. no interpolations or utes) you should hardly hear any difference between different
akes of CD players or between different pressings of the sa e CD. Also „accessories“ like disk
da pening devices or extre ely expensive digital cables will not ake any difference in sonic
quality. Of course it is always a good idea to have a good quality cable for digital (or analog) audio
trans ission - but within reason.
Upsampling, Oversampling and Sampling Rate Conversion in
eneral
In consu er audio circles the two ter s oversa pling and upsa pling are in co on use. Both
ter s essentially ean the sa e, a change in the sa pling frequency to higher values.
Upsa pling usually eans the change in sa pling rate using a dedicated algorith (e.g.
i ple ented on a Digital Signal Processor chip (DSP)) ahead of the final D/A conversion (the D/A
chip), while oversa pling eans the change in sa pling rate e ployed in today’s odern D/A
converter chips the selves.
But let’s start at the beginning. What is the sa pling frequency? For any digital storage or
trans ission it is necessary to have ti e discrete sa ples of the signal which has to be processed.
I.e. the analog signal has to be sa pled at discrete ti e intervals and later converted to digital
nu bers. (Also see "Jitter Suppression and Clocking" above)). This sa pling and conversion
process happens in the so called Analog to Digital Converter (A/D). The inverse in the Digital to
Analog Converter (D/A).

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A physical law states that in order to represent any given analog signal in the digital do ain, one
has to sa ple that signal with at least twice the frequency of the highest frequency contained in
the analog signal. If this law is violated so called aliasing co ponents are generated which are
perceived as a very nasty kind of distortion. So if one defines the audio band of interest to lie
between 0 and 20 kHz, then the ini u sa pling frequency for such signals ust be 40kHz.
For practical reasons explained below, the sa pling frequency of 44.1kHz was chosen for the CD.
A sa pling frequency of 44.1kHz allows to represent signals up to 22.05kHz. The designer of the
syste has to take care that any frequencies above 22.05kHz are sufficiently suppressed before
sa pling at 44.1kHz. This suppression is done with the help of a low pass filter which cuts off the
frequencies above 22.05kHz. In practice such a filter has a li ited steepness, i.e. if it suppresses
frequencies above 22.05kHz it also suppresses frequencies between 20kHz and 22.05kHz to so e
extent. So in order to have a filter which sufficiently suppresses frequencies above 22.05kHz one
has to allow it to have a so called transition band between 20kHz and 22.05kHz where it gradually
builds up its suppression.
Note that so far we have talked about the so called anti-aliasing filter which filters the audio signal
ahead of the A/D conversion process. For the D/A conversion, which is of ore interest to the
High-End Hi-Fi enthusiast, essentially the sa e filter is required. This is because after the D/A
conversion we have a ti e discrete analog signal, i.e. a signal which looks like steps, having the
rate of the sa pling frequency.
Such a signal contains not only the original audio signal between 0 and 20kHz but also replicas of
the sa e signal sy etrical around ultiples of the sa pling frequency. This ay sound
co plicated, but the essence is that there are now signals above 22.05kHz. These signals co e
fro the sa pling process. There are now frequencies above 22.05kHz which have to be
suppressed, so that they do not cause any inter odulation distortion in the a plifier and speakers,
do not burn tweeters or do not ake the dog go ad.
Again, a low pass filter, which is called a „reconstruction filter“, is here to suppress those
frequencies. The sa e applies to the reconstruction filter as to the anti-aliasing filter: Pass-band
up to 20kHz, transisition-band between 20kHz and 22.05kHz, stop-band above 22.05kHz. You ay
think that such a filter is rather "steep", e.g. frequencies between 0 and 20kHz go through
unaffected and frequencies above 22.05kHz are suppressed to aybe 1/100'000th of their initial
value. You are right, such a filter is very steep and as such has so e nasty side effects.
For instance it does strange things to the phase near the cutoff frequency (20kHz) or it shows

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ringing due to the high steepness. In the early days of digital audio these side effects have been
recognized as beeing one of the ain culprits for digital audio to sound bad.
So engineers looked for ways to enhance those filters. They can’t be eli inated because we are
talking laws of physics here. But what if we run the whole thing at higher sa pling rates? Like
96kHz or so? With 96kHz we can allow frequencies up to 48kHz, so the reconstruction filter can
have a transition band between 20kHz and 48kHz, a very uch relaxed frequency response
indeed. So let’s run the whole at 96kHz or even higher! Well – the CD stays at 44.1kHz. So in
order to have that analog lowpass filter (the reconstruction filter) to run at a relaxed frequency
response we have to change the sa pling frequency before the D/A process. Here is where the
Upsa pler co es in. It takes the 44.1kHz fro the CD and upsa ples it to 88.2kHz or 176.4kHz
or even higher. The output of the upsa pler is then fed to the D/A converters which in turn feeds
the reconstruction filter.
All odern audio D/A converter chips have such an upsa pler (or oversa pler) already built into
the chip. One particular chip, for instance, upsa ples the signal by a factor of eight, i.e. 44.1kHz
ends up at 352.8kHz. Such a high sa pling frequency relaxes the job of the reconstruction filter
very uch, it can be built with a si ple 3
rd
order filter.
So, how co e that upsa plers are such a big thing in High-End Hi-Fi circles? The proble with the
upsa plers is that they are filters again, digital ones, but still filters. So in essence the proble of
the analog reconstruction filter has been transferred to the digital do ain into the upsa pler
filters. The big advantage when doing it in the digital do ain is that it can be done with a linear
phase response, which eans that there are no strange phase shifts near 20kHz and the ringing
can also be controlled to so e extent. Digital filters in turn have other proble s and of course
have quite a few degrees of freedo for the designer to specifiy. This eans that the quality of
digital filters can vary at least as uch as the quality of analog filters can. So for a High-End Hi-Fi
designer it is a question whether the oversa pling filter built into the D/A chips lives up to his/her
expectations. If not, he/she can chose to design his/her own upsa pler and bypass part of or the
whole oversa pler in the D/A chip. This gives the High-End Hi-Fi designer yet another degree of
freedo to opti ize the sonic quality of the product.
For the MEDEA+ we have decided to do part of the upsa pling (the ost critical part in fact) in the
Digital Signal Processor (DSP) chip external to the D/A chip.

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Reconstruction Filters
Reconstruction filters have been entioned in the "Upsa pling, Oversa pling and Sa pling Rate
Conversion in General" paragraph above. If you have read that paragraph you know what the
purpose of the reconstruction filter is. The ain point about this analog filter is that its frequency
response should be as s ooth and flat as possible in order to have a virtually linear phase
response. The MEDEA+ e ploys a 3
rd
order filter for that purpose.
Analog Output Stages
The MEDEA+ e ploys class A output stages with a very low output i pedance. Class A inherently
guarantees very low distortion figures even when operating the a plifier in open loop, i.e. without
feedback. The feedback is added to lower the distortion figures even further. The distortion figures
of the MEDEA+, which by the way are excellent, are predo inantely caused by the D/A converter
and not by the output stage.
A very low output i pedance assures that the perfor ance of the MEDEA+ and the subsequent
a plifier co bination is not co pro ised by the cables between the two or by the input
i pedance characteristics of the a plifier. A low output i pedance can be difficult when it co es
to stability issues, but we have taken care of that proble in the MEDEA+.

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Dithering
You have probably not heard the ter dithering in conjunction with audio. Actually it is a ter
widely used in the professional audio real but not so uch in the High-End Hi-Fi arket.
What is dithering? Suppose a digital recording has been ade with a 24 bit A/D converter and a 24
bit recorder. Now this recording should be transferred to a CD which has just 16 bits per sa ple,
as you know. What to do with those 8 bits which are too any? The si plest way is to cut the
off, truncate the . This, unfortunately, generates har onic distortions at low levels, but which
nonethless cause the audio to sound harsh and unpleasant. The har onic distortion is generated
because the eight bits which are cut off fro the 24 bits are correlated with the audio signal, hence
the resulting error is also correlated and thus there are distortions and not just noise (noise would
be uncorrelated). The dithering technique now is used to de-correlate the error fro the signal.
This can be achieved by adding a very low level noise to the original 24 bit signal before
truncation. After truncation the signal does not show any distortion co ponents but a slightly
increased noise floor. This works like agic..... the distortion is replaced by a s all noise – uch
ore pleasant.
I have given the exa ple of a 24 bit recording which has to be truncated to 16 bits. Where is the
application in High-End Hi-Fi audio? More and ore signal processing is i ple ented in the digital
do ain. Think of digital equalizers, digital volu e controls, upsa plers, digital pre-a plifiers,
decoders for encoded signals on DVD etc. All those applications perfor so e athe atical
operations on the digital audio signal. This in turn causes the wordlength of the signal to be
increased. E.g. an input signal to an upsa pler ay have a wordlength of 16 bits (off a CD), but
the output signal of the upsa pler ay have 24 bits or even ore. This co es fro the fact that
the athe atical operations e ployed in such devices increase the word length. E.g. a
ultiplication of two 2 digit nu bers results in a four digit nu ber. So after the upsa pler the
word length ay be higher than the subsequent processor ay be able to accept. In this exa ple,
after the upsa pler there ay be a D/A converter with a 24 bit input word length capability. So if
the upsa pler generates a word length of ore than 24 bits it should be dithered to 24 bits for
axi u signal fidelity.

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I hope these excursions into the theory and practice of audio engineering have been useful for you.
If you would like to dive further into those issues I reco end to visit the website of Mr. Bob Katz,
a renowned Mastering Engineer and a Weiss Engineering custo er. He publishes articles on
Dithering and Jitter and any other topics at http://www.digido.co /

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THE MEDEA+ D/A CONVERTER
Features
Inputs:
There are three AES/EBU inputs on XLR connectors, and one on Toslink (optical). In addition each of the four inputs has an
S/PDIF (RCA) connector. The accepted sa pling frequencies are 44.1, 48, 88.2, 96, 176.4 or 192 kHz. Sa pling frequencies of
88.2kHz and higher can also be fed using the dual wire sche e, i.e. one cable per channel.
Upsampling:
Signals at 44.1 or 48 kHz are upsa pled in the MEDEA+’s DSP chip, using our own upsa pling algorith s. Thus the DAC chip
always receives signals at a 88.2kHz ini u sa pling rate.
Synchronization:
Several signal reclocking sche es are co bined for extre ely high jitter attenuation, aking the MEDEA+ virtually i une to
jitter over a very wide bandwidth.
Converters:
The DAC chip upsa ples the input signal to about 1.5MHz. Four D/A conversion channels are used per audio channel, resulting
in exceptional perfor ance specifications. The I/V conversion after the DAC chip uses two of our own discrete OP-AMPs (the
OP1-BP) per channel.

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Outputs:
Our discrete OP-AMPs are also used in the output stage, two per audio channel for a differential output. Output levels can be
set between –4.0dBu and +27.0dBu (6dB less on asy etrical outputs). The level setting is done in 1dB steps via a rotary
switch at the back of the unit. Both outputs are set to the sa e level at any ti e. The outputs are sy etrical on XLR
connectors and asy etrical on RCA connectors. No sound degrading servo echanis s are used.
The Weiss OP1-BP discrete OP-AMP
Power Supply:
A powerful non-switching power supply is used. All sensitive voltages have their own regulators which are separated between
left and right channels. The result is an analog output free of "digital noise" and channel crosstalk. The power switch activates
a se iconductor relais which only switches on or off at zero crossings of the ains voltage. This assures a glitch free power
switching.

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Mechanical Design:
The MEDEA+ e ploys a twin etal fra e. The inner chassis is ade of steel and acts as a very effective shield against
electrostatic and electro agnetic radiation. The outer fra e is ade of assive anodized alu iniu for additional shielding,
resulting in opti al heat convection and beautiful looks.
Frontpanel controls:
• Four switches with corresponding blue LEDs are used for selecting one out of four input sources.
• A power switch.
Backpanel elements from left to right:
• Analog outputs on XLR and RCA connectors with output level switch.
• Digital inputs 1 – 4 (three XLR, four RCA and one Toslink connector).
• Mains connector with fuses and voltage selector.

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Operation
Unpacking and Setup of the MEDEA+
Carfully unpack the MEDEA+. The following ite s should be enclosed:
• The MEDEA+ D/A Converter unit
• This Owners Manual
• A Certificate of Guarantee
The MEDEA+ should be placed horizontally on a plane surface. Avoid putting any other equip ent
or any other aterial on top of the MEDEA+ as not to co pro ise the MEDEA+’s heat convection
syste .
Mains Connection
Before connecting the ains cable ake sure the voltage selector switch on the back of the unit
(integrated into the ains connector) is set to the appropriate voltage. If this is not the case then
open the ains selector lid with a strong screwdriver. The ains plug ust not be plugged into the
connector for that purpose. Now you see a black selector dru with appropriate voltages written in
white letters. Re ove this dru , turn it to the appropriate voltage and reinsert it. Now fir ly close
the lid again. Connect the MEDEA+ to the wall outlet with the ains cable.

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Input Source Connection, Selection and Synchronization
Connect sources with sa pling frequencies of 44.1, 48, 88.2, 96, 176.4 or 192kHz to the input
connectors at the back of the unit. Note that there are a total of four inputs. The following
connectors are available:
Input # 1: one XLR and one RCA connector (use either XLR or RCA at a ti e, i.e. you ay connect
two sources si ultaneously, but only one source unit ust be switched on at a ti e).
Input # 2: The sa e as Input # 1 applies here.
Input # 3: The sa e as Input # 1 applies here.
Input # 4: one TOSLINK (optical) and one RCA connector (use either TOSLINK or RCA at a ti e,
i.e. you ay connect two sources si ultaneously, but only one source unit ust be switched on at
a ti e).
The input source pushbuttons correspond to the four inputs as described above.
Input # 1 (left) and Input #2 (right) can act as an input for sources supporting the two wire
sche e, i.e. one cable per channel. The MEDEA+ auto atically detects whenever such a source is
connected to inputs #1 and #2 and switches to the appropriate conversion ode. In that ode
both LEDs of input #1 and #2 are lit. The two wire sche e is used in so e sources with sa pling
rates of 88.2kHz or higher.
If a valid AES/EBU or S/PDIF for atted signal is present at the selected input, the blue LED stays
lit. If the for at and/or the sa pling frequency is not valid, the blue LED is flashing. In that case
the analog outputs are uted.
The MEDEA+ auto atically synchronizes to the inco ing signal. If the sa pling rate of the signal
is not within a range of +- 80pp of one of the no inal sa pling rates (44.1, 48, 88.2, 96, 172.4,
192kHz) then the blue LED of the active input is flashing. The analog outputs are uted in that
case.
The states of all buttons are stored in non-volatile e ory, i.e. the actual input selection is
retained when powering the unit off.

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Output Level and Output Connection
Connect your prea plifier or power a plifier to the output connectors of the MEDEA+. Use either
the sy etrical (balanced) lines (XLR) or the asy etrical (unbalanced) lines (RCA). Or both, if
you need to feed two a plifiers. Sy etrical is preferred, provided your prea plifier supports
sy etrical connection.
The output level is set via the rotary switch on the backpanel. The position of that switch does not
indicate the output level. The switch erely acts as a rotary encoder, i.e. rotating clockwise
increases the level and vice versa. Adjust that switch to the proper output level which atches
your a plifier. If your a plifier has a volu e control, adjust the MEDEA+ output such that you get
a nor al listening level with the a plifier’s volu e control set to about idway between ini u
and axi u volu e.
Note, it is nor al when after setting the rotary volu e switch the MEDEA+ output utes again for
a short ti e.