WEISS DAC202 User manual

Weiss Engineering Ltd.
Florastrasse 42, 8610 Uster, Switzerland
www.weiss-highend.co
DAC202
OWNERS MANUAL

OWNERS MANUAL FOR WEISS DAC202 D/A CONVERTER
Page 2 Date: 03/10
INTRODUCTION
Dear custo er
Congratulations on your purchase of the DAC202 D/A Converter and
welco e to the fa ily of Weiss equip ent owners!
The DAC202 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 DAC202 converter.
I wish you a long-lasting relationship with your DAC202.
Yours sincerely,
Daniel Weiss
President, Weiss Engineering Ltd.

OWNERS MANUAL FOR WEISS DAC202 D/A CONVERTER
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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
7 Upsa pling, Oversa pling and Sa pling Rate
Conversion in General
9 Reconstruction Filters
10 Analog Output Stages
10 Dithering
12 The DAC202 D/A Converter
12 Features in alphabetical order
15 Operation, Installation
26 Software Installation
27 Software Setup
29 Technical Data
32 Contact

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A SHORT HISTORY OF WEISS ENGINEERING
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 RODUCT HILOSO HY
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 and the
VESTA Firewire – AES/EBU Interface
2010 Introduction of the DAC202 Firewire DAC, the
INT202 Firewire Interface and the ATT202
Passive Attenuator

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ADVANCED DIGITAL AND ANALOG AUDIO
CONCE TS EX LAINED
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“.
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.

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In the standalone D/A converter jitter can be introduced
by inferior cables between the source (e.g. CD transport)
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
DAC202), 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 (e.g. 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 flywheel.
In the DAC202 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 DAC202 suppresses even
very low frequency jitter co ponents.
This eans that the DAC202 is virtually i une to the
quality of the audio source regarding jitter. For a CD
transport 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 transports 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
General
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

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(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).
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.

OWNERS MANUAL FOR WEISS DAC202 D/A CONVERTER
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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 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.
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

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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 DAC202 e ploys a 3
rd
order filter
for that purpose.
Analog Output Stages
The DAC202 e ploys separate output stages for the
ain output and the headphone output. Both stages use
state of the art operational a plifiers with high slew
rate. A topology with a very low output i pedance has
been chosen. This assures that the perfor ance of the
DAC202 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.
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

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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.
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 DAC202 D/A CONVERTER
Features in alphabetical order
Absolute polarity switch:
The absolute polarity of the outputs can be inverted for
opti izing the sonic i pression.
Audio Inputs:
One XLR, one RCA and one Toslink connector for
AES/EBU or S/PDIF signals. Two Firewire connectors for
co puter connection.
Audio Outputs:
Two XLR and two RCA connectors for analog audio
output. One XLR and one RCA connector for AES/EBU
and S/PDIF audio output. One ¼ inch jack socket for
headphones.
Backpanel elements from left to right:
• Analog outputs on XLR and RCA connectors
• Digital outputs on XLR and RCA connectors
• Digital inputs on XLR and RCA connectors
• Wordsync input and output on BNC connectors
• Digital input on Toslink connector
• Firewire connectors
• Mains connector with fuse
Converters:
Two converters per channel are e ployed in order to
lower the converter i perfections. Separate converters
are used for the ain outputs and the headphone
outputs.

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Dual / Single Wire modes:
The AES/EBU inputs / outputs of the DAC202 nor ally
work in the so called “Single Wire” odus, i.e. both
audio channels are transferred via a single cable. The
DAC202 also supports the “Dual Wire” odus where the
two audio channels are transferred via two cables, i.e.
left channel is on the XLR connector and the right
channel on the RCA connector. This applies to both input
and output connectors. The Dual Wire odus, when
activated, is active only at sa pling rates of 176.4 or
192 kHz.
In Dual Wire ode the frequency of the wordclock
synchronization on the BNC connectors can be chosen to
be the sa pling rate (i.e. 176.4 or 192 kHz) or half the
sa pling rate (i.e. 88.2 or 96 kHz).
Frontpanel elements:
• Standby LED
• IR Receiver
• Headphone socket
• LCD display
• Rotary encoder with switch
Insert mode:
If the insert ode is activated an external digital audio
device (e.g. a digital equalizer) can be looped into the
signal path via the XLR input / output connectors. The
resulting signal path thus looks as follows: Firewire (or
RCA or Toslink) input  XLR output  external device 
XLR input  DAC chip.
LCD brightness:
The brightness of the LCD can be set with two different
choices: One brightness level is active when operating
the rotary encoder knob or the re ote control. The other
brightness level is active when the DAC202 or the
re ote control are not touched. This allows to di the
LCD when the infor ation on the LCD screen is not
required.
Level Control Main Output:
The ouput level of the ain output can be adjusted in
the analog do ain in four coarse steps in order to

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Page 14 Date: 03/10
acco odate for the input sensitivity of the subsequent
a plifier. A higher resolution level control is
i ple ented in the digital do ain and operated fro the
frontpanel knob or the re ote control. The high
resolution level control can be defeated for the ain
output in case there is another level control available in
the audio chain.
Level Control Headphone Output:
The ouput level of the headphone output can be adjusted
in the analog do ain in four coarse steps in order to
acco odate for the headphone sensitivity. A higher
resolution level control is i ple ented in the digital
do ain and operated fro the frontpanel knob or the
re ote control.
ower Supply:
A powerful non-switching power supply is used. All
sensitive voltages have their own regulators.
Remote Control:
The IR re ote control allows to control the
following para eters:
- Power on / off
- Volu e up / down
- Input source (Firewire, XLR, RCA,
Toslink)
- Output ute
- Polarity nor al / inverted
- Upsa pling filter type
Signal routing:
Due to the various possible settings for input source,
insert ode, dual/single wire odes and sync source
there are quite a few routing pathes possible. Refer to
the operation instructions below for a detailed list of the
signal routing.
Synchronization:
Wordclock input and output on BNC connectors.
Supported sa pling rates on all inputs: 44.1, 48, 88.2,
96, 176.4, 192 kHz. For all input odes the
synchronization source can be freely chosen. E.g. with
Firewire as input the “Internal” synchronization is
typically chosen, which eans that the DAC202 is the
aster clock for the co puter.

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Transparency check:
The bit transparency of any player software running on a
co puter can be verified by this feature. For that
purpose audio files are supplied with the DAC202.
Playing back these files via the player software to be
checked allows the DAC202 software to recognize the bit
pattern of the files. If the bits of the files are changed
during playback e.g. because of a volu e control or EQ
or upsa pling algorith etc., the bit transparency check
fails. The files supplied cover all the DAC202 supported
sa pling rates (44.1, 48, 88.2, 96, 176.4, 192 kHz) as
well as 16 and 24 bit wordlengths.
Upsampling Filter selection:
The upsa pling filter can be selected bewen “A” and “B”.
Filter A has a steeper frequency response than B. Future
DAC202 software will offer ore filter choices. All
DAC202 units can be software updated via Firewire.
Operation / Installation
Unpacking and Setup of the DAC202
Carefully unpack the DAC202. The following ite s should
be enclosed:
• The DAC202 D/A Converter unit
• The IR re ote control unit
• A CD with the necessary Firewire drivers for
Windows and OSX and with the audio files for the bit
transparency check
• This Owners Manual
• A Certificate of Guarantee
Firewire Connection
Before connecting the firewire cable between co puter
and DAC202 unplug both the co puter and the DAC202
fro the ains power.
Mains Connection
Before connecting the ains cable ake sure the label
on the back of the unit (near the ains inlet) shows the
appropriate ains voltage. If this is not the case then
the proper ains voltage ay have to be selected with a
ju per cable inside of the DAC202 unit. Contact your
dealer in that case.

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First time operation
After connecting the necessary cables (the DAC202 can
also be operated without co puter, e.g. by connecting a
CD transport to one of its inputs) switch on the unit by
pressing on the rotary encoder knob.
Note that when power is applied to the DAC202 the
blue standby LED is lit. When the DAC202 is
switched on, the blue LED is turned off and after a
short while the LCD screen comes on.
Note that if a co puter is connected to the DAC202 via
Firewire, the sync source and sync frequency (if
applicable) have to be selected fro within the Weiss
Firewire IO window on the co puter. The selection fro
the DAC202 screen does not work in that case!
After a short while the LCD screen lits up and shows the
basic start-up screen, e.g. like shown here.
In the upper left corner the volu e in dB (deciBel) is
shown, a value of 0.0 is axi u volu e. Below the dB
figure there is a bar which also represents the volu e.
In the upper right corner the polarity is shown with the
Greek character “phi” which is used for the phase angle
in electrical engineering. “Phi +” eans the signal is not
inverted, “phi –“ eans the signal is inverted (both
channels).
In the lower right corner the selected upsa pling filter
type is shown. It can be “A” or “B” and for later software
versions it ay be even “C”, “D” etc.
Below the volu e bar the current input source is shown.
It can be “Firewire”, “AES (XLR)”, “SPDIF (RCA)” or
“SPDIF (TOS)”. These are the four input sockets to the
DAC202, i.e. Firewire, XLR, RCA and Toslink (optical).
Below the input source the sa pling rate is shown, if
there is a valid signal present at the selected input,
otherwise “unlocked” indicates that there isn’t a valid
input signal.
Rotating the knob causes the volu e control to change
the value.
Pressing the knob when the display shows the ain
enu activates the selection of the “Options Menu” as
shown here.

OWNERS MANUAL FOR WEISS DAC202 D/A CONVERTER
Page 17 Date: 03/10
Pressing the knob again enters the “Options Menu”.
Rotating the knob instead of pressing it navigates to the
input source select as shown here. Pressing the knob
again allows to select the input source. If any para eter
is shown with the two dots in the upper left and lower
left corners, then that para eter can be changed by
rotating the knob. To confir a setting the knob has to
be pressed again. The two dots vanish and the
para eter is set to the value indicated. This picture
shows the two dots in the input selection enu.
Selectable input sources are:
1. FireWire
2. AES (XLR)
3. SPDIF (RCA)
4. SPDIF (TOSLINK) –note: up to 48kHz sa pling rate
only!
This picture shows the AES/EBU input on XLR selected.
There isn’t any valid signal at the AES/EBU input thus it
shows “unlocked”.
The “Options Menu” explained:
Upon entering the Options Menus the display as shown
here appears. The highlighted ite is the one which can
be changed / executed by pressing the knob. E.g. if the
knob is pressed with the display as shown, the Options
Menu is exited.
Here is a rundown of all entries in the Options Menu:
1.) Abs. Phase: “+” or “-“. A “+” eans the signal is
not inverted when passing through the DAC202,
A “-“ eans the signal is inverted.
2.) Upsa ple Filt.: upsa ple filter type “A” or “B”.
Later software versions ay allow to select “C”,
“D” etc. “A” uses a steeper filter than “B”. Also
see the Technical Data section.
3.) Sync Source: (these instructions assu e that
neither dual wire nor insert odes are selected,
for dual wire and/or insert odes check the
instructions further down.)
Note that if a computer is connected to the
DAC202 via Firewire, the Sync Source
parameter has to be selected from the
Weiss Firewire IO control panel on the
computer!

OWNERS MANUAL FOR WEISS DAC202 D/A CONVERTER
Page 18 Date: 03/10
For all possible input sources (Firewire,
AES(XLR), SPDIF(RCA) and SPDIF(TOS)) the
following sync sources can be selected:
- XLR: This selects the XLR input as the
synchronization source.
- RCA: This selects the RCA input as the
synchronization source.
- Toslink: This selects the Toslink input as the
synchronization source.
- WC BNC: This selects the BNC connector at the
rear of the DAC202 as the synchronization
source. If the DAC202 is used in dual wire ode
read the instructions for the dual wire ode
regarding external synchronization.
- 1394 bus: This slaves the DAC202 clock to the
Firewire bus. This setting is only required if ore
than one DAC202 unit is connected to the sa e
co puter for ultichannel playback. In that case
one of the DAC202 is the aster clock and the
other DAC202 units have to be slaved to that
aster DAC202. This is done by setting the slave
DAC202 to “1394 bus”.
- Internal: The DAC202 generates the sa pling
rate clock internally. Note that in this ode the
source has to be synchronized to the internally
generated sync. With Firewire as input source
this is done auto atically via Firewire. With the
other inputs the source, e.g. a CD transport has
to by synchronized via e.g. the sync out
BNC connector at the back of the DAC202.
4.) Sync Rate: Depending on the Sync Source
selected there is either the sa pling rate shown
or the word “autolock”. If the sa pling rate is
indicated (i.e. “Internal” is selected as Sync
Source) then it is possible to change the rate to
the appropriate value. Usually the sa pling rate
is set by the player progra running on the
co puter in that case.
5.) LCD Bright: sets the LCD brightness
6.) LCD Di Lev.: sets the LCD brightness when in
di ed ode. The di ed ode is entered
after so e ti e where there weren’t done any
changes to the para eters via the frontpanel
knob or the re ote control.
7.) Dual Wire: Disabled eans that the signals at
the XLR or RCA or Toslink inputs are treated as
single wire AES/EBU signals with sa pling rates
up to 192kHz. Also the XLR and RCA outputs
operate in single wire ode up to 192kHz. If
enabled, the XLR and RCA inputs (or outputs)
are a dual wire pair, i.e. the XLR connectors
carry the left channel and the RCA connectors
carry the right channel. This is the case only for
sa pling rates of 176.4 or 192 kHz though! All
other sa pling rates continue to work in single
wire ode.
8.) DW WCLK: Means Dual Wire Wordclock. Can be
audiorate or halfrate. Audiorate eans that the

OWNERS MANUAL FOR WEISS DAC202 D/A CONVERTER
Page 19 Date: 03/10
wordclock signal at the BNC connectors (input or
output) is at the actual audio sa pling rate when
the unit operates in dual wire ode. I.e. the
wordclock rate at the BNC connectors is:
Audio Sa pling rate: BNC connectors rate:
44.1 44.1
48 48
88.2 88.2
96 96
176.4 176.4
192 192
If “halfrate” is selected the wordclock rate at the
BNC connectors is:
Audio Sa pling rate: BNC connectors rate:
44.1 44.1
48 48
88.2 88.2
96 96
176.4 88.2
192 96
9.) Insert Mode: When enabled, an external digital
audio device (e.g. a digital equalizer) can be
inserted into the signal path between e.g. the
source via Firewire and the D/A converter. The
insert ode possibilities are explained further
down.
10.) Main Out Att.: If engaged, the volu e knob
and the re ote control volu e work on both the
volu e of the ain output and the headphone
output. This ode is selected if the DAC202 is
used as a prea plifier. If bypassed, the volu e
knob and the re ote control volu e work only
on the headphone output. The ain outputs are
set to full volu e (0.0 dB). This ode is used if
there is another volu e control used in the
chain.
11.) XLR Out Lev.: Main output level in Vr s. There
are four settings to chose fro : 8.15Vr s,
4.15Vr s, 2.12Vr s and 1.06Vr s. Best is to
start off with the lowest value (1.06V) and have
the volu e knob at 0.0. If the audio volu e is at
a co fortable level, i.e. does not need to be
louder, leave the setting at 1.06V. If it needs to
be louder select the next setting (2.12V). I.e.
select the setting which gives you a co fortably
loud level with the volu e knob set to 0.0, i.e.
the axi u level.
12.) Phones Lev.: The sa e as the Main Out Level,
but for the headphone output. Be careful when
selecting that level! The settings are: 0.2Vr s,
0.9Vr s, 2.7Vr s, 5.2Vr s. Start off with the
lowest level (0.2Vr s). This level is fine for
any low i pedance headphones. If the volu e
is too low even for a 0.0 setting of the volu e
knob then get to the next higher setting. The
highest setting (5.2Vr s) is used for very
insensitive headphones like e.g. the AKG K1000.

OWNERS MANUAL FOR WEISS DAC202 D/A CONVERTER
Page 20 Date: 03/10
insensitive headphones like e.g. the AKG K1000.
13.) Transparency: This allows to check the player
progra on your co puter for bit transparency.
To do this you need to play the audio files
supplied on the CD co ing with the DAC202.
Copy these files onto your drive holding your
audio files. There are two files for each sa pling
rate, one at a 16 Bit wordlength (I.e. the syste
is checked for 16 bit transparency) and one at a
24 Bit wordlenght for 24 bit transparency
checking. The files are in WAV for at which is an
unco pressed for at supported by ost
players. When playing a particular file ake sure
the DAC202 shows the sa e sa pling rate as
the file played has. If the two rates do not atch
then there is a sa pling rate conversion going
on and bit transparency can not be achieved.
When this is all fine, play the file and activate
the Transparency check by pressing the button
when the “run” word is highlighted. If the player
software is bit transparent then the wordlength
of the file played is shown, i.e. 16Bit or 24Bit. If
the player software is not bit transparent the
word “fail” is shown. “fail” eans that the bits of
the original audio file get changed so ewhere on
the path between the harddisk and the DAC202.
Places to look for bit changes are: volu e
controls, equalizers, “sound enhancers”,
sa pling rate conversions, ixers.
Note that the test audio files do not generate any
audible audio signal. This akes sure that your
speakers are protected when doing the test.
14.) Syste Info: Infor ation on the operating
syste version etc.