WEISS INT202 User manual

Weiss Engineering Ltd.
Florastrasse 42, 8610 Uster, Switzerland
www.weiss-highend.co
INT202
OWNERS MANUAL

OPERATING INSTRUCTIONS FOR THE INT202 FIREWIRE INTERFACE
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INTRODUCTION
Dear Custo er
Congratulations on your purchase of the INT202 Firewire
Interface and welco e to the fa ily of Weiss equip ent
owners!
On the following pages I will introduce you to our views
on high quality audio processing and interfacing. These
include funda ental digital and analog audio concepts
and the INT202 Firewire Interface.
I wish you a long-lasting relationship with your INT202.
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
10 Dithering
11 Firewire vs USB
13 The INT202 Firewire Interface
13 Features
16 Operation / Installation
21 Technical Data
22 Contact
Author: Daniel Weiss, Weiss Engineering LTD.
Weiss Engineering LTD. reserves the right to ake changes to product specification or docu entation without prior
notice. Updated anuals and datasheets are available at our website for downloading. Weiss Engineering LTD. akes
no warranty, representation or guarantee regarding the suitability of its products for any particular purpose, nor does
part of this anual, and specifically discali s any and all liability, including without li itation consequential or
incidental da ages.
All rights reserved. No part of this publication ay be reproduced or trans itted in any for or by any eans,
electronic or echanical including photocopying, scanning or any infor ation storage or retrieval syste without the
express prior written consent of the publisher.
© Copyright Weiss Engineering LTD., 2010.

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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,
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 ore than
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.
The e are ome of the mile tone at Wei
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
2009 Introduction of the INT202 Firewire Interface, the
ATT202 Passive Attenuator, the DAC202 D/A
Converter

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ADVANCED DIGITAL AND ANALOG AUDIO
CONCEPTS EXPLAINED
Jitter Suppre ion 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

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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
MINERVA), 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.
Jitter handling in the INT202 Firewire Interface in
more detail
The Jitter Eli ination Technologies (JET) PLL on the chip
used in the INT202 feature state-of-the art jitter
rejection abilities and extre ely low intrinsic jitter levels.
Like all phase-locked loops, JET PLL use feedback to lock
an oscillator to a ti ing reference. They track slow
reference changes, but effectively free-run through rapid
odulations of the reference (i.e. flywheel like). Fro a
jitter transfer point of view, they provide increasing jitter
attenuation above so e chosen corner frequency.
Jitter attenuation is just one aspect of PLL design. Other
considerations include frequency range and intrinsic
jitter. It can be shown that conventional designs are
bound by a funda ental tradeoff between these three
aspects. For exa ple, specifying a frequency range of
one octave eans using a low-Q oscillator. But that
akes for high intrinsic jitter when the loop corner
frequency is held down. Conversely, good jitter
attenuation and low intrinsic jitter can be had by using a
voltage-controlled crystal oscillator (VCXO). But the
frequency range is then tiny. A further consideration is
that only low-Q oscillators are easy to integrate on chip.
JET PLL sidestep the above- entioned tradeoff. It
incorporates two loops. One is largely or wholly nu eric,

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and has its corner frequency set low enough to give good
reference-jitter attenuation. The other regulates the
analog oscillator and has its corner frequency set uch
higher, to oderate the intrinsic jitter. The two corner
frequencies ight be around 10 Hz and 100 kHz, for
exa ple. Another benefit of having a high corner
frequency in the analog loop is that interference, e.g. via
the oscillator's supply rail, is ore-effectively
suppressed. JET PLL requires a fast, stable, fixed-
frequency clock. It is this that gives it stability in the
band between the two corner frequencies. (Equally, in
this band any jitter on this clock passes straight through
to the JET PLL's clock output.) The stable clock is usually
derived fro a free-running crystal oscillator. JET PLL
contains a nu ber-controlled oscillator, which can also
be called a fractional frequency divider. Like the analog
oscillator, this injects jitter. Typically, spectru shaping
is used to push ost of that jitter up to frequencies
where it will be heavily attenuated by the analog loop. As
well as frequency-locking the analog oscillator to the
provided reference, JET PLL can also phase-lock an
associated fra e sync to the reference.
Up ampling, Over ampling and
Sampling Rate Conver ion 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
(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

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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 i 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

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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.
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.