Weiss Engineering INT204 Owner's Manual

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INT204

Owner’s Manual

Weiss Engineering ltd.

Florastrasse 42

8610 Uster

Switzerland

Phone: +41 44 940 20 06

Fax: +41 44 940 22 14

Email: [email protected]

Web: www.weiss.ch

www.asiaweiss.com

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iii

Introduction

Dear customer,

Congratulations on your purchase of the INT204 USB/DSD Interface and welcome to the family of Weiss equipment owners!

On the following pages I will introduce you to our views on high quality audio processing. These include fundamental digital and analog audio concepts and the INT204 USB/DSD Interface.

I wish you a long-lasting relationship with your INT204.

Yours sincerely, Daniel Weiss

President, Weiss Engineering lt d.

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Authors: Daniel Weiss, Weiss Engineering lt d.

TEXLayout: Samuel Groner & Rolf Anderegg, Weiss Engineering ltd .

Date: June 30, 2017

Weiss Engineering ltd . reserves the right to make changes to product speciﬁca-

tion or documentation without prior notice. Updated manuals and datasheets are available at our website for downloading. Weiss Engineering ltd . makes no

warranty, representation or guarantee regarding the suitability of its products for

any particular purpose, nor does part of this manual, and speciﬁcally disclaims any and all liability, including without limitation consequential or incidental damages. All rights reserved. No part of this publication may be reproduced or transmitted in any form or by any means, electronic or mechanical including photocopying, scanning or any information storage or retrieval system without the express prior written consent of the publisher.

 Copyright Weiss Engineering lt d. , 2017

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A Short History of Weiss

Engineering Ltd.

Having studied electrical engineering, Daniel Weiss joined the Willi Studer

(Studer-Revox) company in Switzerland. His work included the design of a

sampling frequency converter and of digital signal processing electronics for digital audio recorders. In 1985, Mr. Weiss founded the company Weiss Engineering ltd . From the outset the company concentrated on the design and manufacture of digital audio equipment for mastering studios. Its ﬁrst product was the modular 102 Series system. After more than 25 years, this system is still up to date (24 bit/96kHz).

In the early nineties the Gambit Series was launched, taking ergonomics and sonic quality to new heights. The Gambit Series consists of stand-alone units like equalizer, denoiser/declicker, dynamics processor, A/D converter, D/A converter, sampling frequency converter, dithering etc. 40 bit ﬂoating point processors and sampling rates up to 192 kHz are employed. In 2001 we have decided to enter the High-End Hi-Fi market, which oﬀers a comparable clientele to that of the mastering studios. Both consist of critical and discerning listeners, who are in constant search for the best audio reproduction equipment or the best audio tools respectively.

Our list of clients includes big names, like SONY, BMG, EMI, Warner, Hit

Factory, Abbey Road, Teldec, Telarc, Gateway Mastering (Bob Ludwig), Bernie Grundman Mastering, Masterdisk, Sterling Sound, Whitﬁeld Street, Metropolis

and hundreds more. For a more comprehensive list you are invited to visit our pro audio website at www.weiss.ch.

Today Weiss Engineering ltd . employs nine people, ﬁve of them in the engineering department.

Our Mission and Product Philosophy

The wealth of experience we have gained in over 25 years of designing products

for top mastering engineers, we now also apply to the design of outstanding High-End Hi-Fi products. Our mission is to create equipment which becomes classic right from the outset — outstanding in sonics and design. These are some of the milestones at Weiss Engineering ltd .:

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1985

Introduction of the 102 Series, a 24 bit modular digital audio processor for

mastering studios

1986 Introduction of one of the ﬁrst sample rate converters for digital audio

1987 Introduction of one of the ﬁrst digital equalizers

1989 Introduction of one of the ﬁrst digital dynamics processors

1991

Introduction of the Ibis digital mixing console, built for the mix-down of

classical music

1993

Introduction of the Gambit Series of digital audio processors, which employ 40 bit ﬂoating point processing and sport an extremely ergonomic user interface

1995 First 96 kHz sampling rate capable products delivered

2001

Introduction of the MEDEA High-End Hi-Fi D/A Converter, the ﬁrst product in our High-End series

2004 Introduction of the JASON CD Transport

2007 Introduction of the CASTOR High-End Hi-Fi Power Ampliﬁer

2008

Introduction of the MINERVA Firewire D/A Converter and the VESTA Firewire-AES/EBU interface

2010

Introduction of the DAC202 Firewire D/A Converter, the INT202 Firewire Interface and the ATT202 Passive Attenuator

2011

Introduction of the MAN301 Music Archive Network Player and the INT203 Firewire I/O Interface

2012

Introduction of the MEDEA+ D/A Converter, the MEDEA with an enhanced analog section

2013

Introduction of the INT204 USB/DSD Interface and the MEDUS State Of The Art D/A Converter.

2017

Introduction of the DAC501/DAC502 DSP D/A Converter and Network Renderer.

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Contents

Introduction iii

A Short History of Weiss Engineering Ltd. v

1 Advanced Digital & Analog Audio Concepts Explained 3

1.1 Jitter Suppression & Clocking . . . . . . . . . . . . . . . . . . . . 3

1.2 Up-/Oversampling & Sampling Rate Conversion in General . . . 4

1.3 Dithering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

1.4 Digital Level Control . . . . . . . . . . . . . . . . . . . . . . . . . 7

2 The INT204 USB/DSD Interface 11

2.1 Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

2.1.1 Input . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

2.1.2 Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

2.1.3 Synchronization . . . . . . . . . . . . . . . . . . . . . . . 11

2.1.4 DSD to PCM conversion . . . . . . . . . . . . . . . . . . . 11

2.1.5 Power Supply . . . . . . . . . . . . . . . . . . . . . . . . . 12

2.1.6 Front Panel . . . . . . . . . . . . . . . . . . . . . . . . . . 12

2.1.7 Back Panel . . . . . . . . . . . . . . . . . . . . . . . . . . 12

2.2 Installation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

2.2.1 Software installation . . . . . . . . . . . . . . . . . . . . . 12

2.3 Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14

2.3.1 USB and Power Connection . . . . . . . . . . . . . . . . . 14

2.3.2 AES/EBU or S/PDIF Connection . . . . . . . . . . . . . 14

2.3.3 DSD to PCM Conversion . . . . . . . . . . . . . . . . . . 14

2.3.4 PCM to PCM Conversion . . . . . . . . . . . . . . . . . . 15

2.3.5 LED Sampling Frequency Indication . . . . . . . . . . . . 15

2.3.6 Optional IR Level Control . . . . . . . . . . . . . . . . . . 15

3 INT204 Technical Data 17

3.1 Digital Inputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

3.2 Digital Outputs . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

3.3 Synchronization . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

3.4 DSD Conversion . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

3.5 Power . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18

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A Some information on the DSD format — properties, caveats,

uses. 19

A.1 Quantizing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

A.2 Filtering . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

A.3 DSD uses . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

A.4 Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

A.5 What is better? . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

A.6 The future . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

B Contact 21

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Chapter 1

Advanced Digital and Analog Audio Concepts

Explained

1.1 Jitter Suppression and Clocking

What is jitter and how does it aﬀect audio quality? In the audio ﬁeld the term

jitter designates a timing uncertainty of digital clock signals. In an analog to digital converter (A/D) the analog signal is sampled (measured) at regular time intervals; in the case of a CD, 44 100 times a second or every 22.675737µs

(microseconds). If these time 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 same time interval between each and every sample. After all, even

digital signals are analog in their properties and thus are inﬂuenced by noise, crosstalk, power supply ﬂuctuations, temperature etc.

Hence a jittery clock introduces errors to the measurements taken by the

A/D, resulting from measurements being taken at the wrong time. 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 form. A good designer takes care that the jitter amount in his/her design is minimized as well as possible.

What type of equipment can be compromised by jitter? There are three

types: The A/D converter as described above, then there is the D/A converter

where the same mechanism as in the A/D converter applies and the third is

the asynchronous sample rate converter (ASRC). The ASRC is not something usually found in Hi-Fi systems. It is used by sound engineers to change the sample rate from e.g. 96 kHz to 44.1 kHz, or e.g. for putting a 96 kHz recording onto a 44.1 kHz CD.

You may now argue that in High-End Hi-Fi there are such things as”over-

samplers“ or”upsamplers“. Yes, those are in essence sampling rate converters,

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however in a well-designed system these converters employ a synchronous design,

where jitter does not play any role. Of course a conversion between 96 kHz and

44.1 kHz as in the example above can be done in a synchronous manner as well. An ASRC in fact is only required either where one or both of the sampling

frequencies involved are changing over time (”varispeed“ mode of digital audio recorders) or where it is unpractical to synchronize the two sampling frequencies.

So basically in Hi-Fi jitter matters where there are A/D or D/A converters involved. CD and DVD players are by far the most numerous type of equipment employing 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 mechanisms, like power supply contamination by power hungry motors

(spindle/servo), microphony of the crystal generating the sampling 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 transport) and the D/A converter unit or by the same mechanisms as described above — except for the motors of course. In the case of a stand-alone D/A converter (as the DAC202), one has to take two diﬀerent jitter contamination paths into account. One is the internal path where internal signals can aﬀect the jitter amount of the sampling clock generator. Here, all the good old analog design principles have to be applied. Such as shielding from electric or magnetic ﬁelds, good grounding, good power supply decoupling, good signal transmission between the clock generator and the actual D/A chip.

The other path is the external signal coming from the source to which the sampling clock has to be locked. In other words the D/A converter has to run synchronous to the incoming digital audio signal and thus the frequency of the internal sampling clock generator has to be controlled so that it runs at the same sampling speed as the source (e.g. CD transport). This controlling is done by a phase locked loop (PLL), which is a control system with error feedback. Of course the PLL has to be able to follow the long term ﬂuctuations of the source, e.g. the sampling rate of the source will alter slightly over time or over temperature, it will not be a constant 44.1 kHz in the case of a CD. But the PLL should not follow the short-term ﬂuctuations (jitter). Think of the PLL as being like a very slow-reacting ﬂywheel.

1.2 Upsampling, Oversampling and Sampling

Rate Conversion in General

In consumer audio circles the two terms oversampling and upsampling are in common use. Both terms essentially mean the same, a change in the sampling frequency to higher values. Upsampling usually means the change in sampling rate using a dedicated algorithm (e.g. implemented on a digital signal processor chip (DSP)) ahead of the ﬁnal D/A conversion (the D/A chip), while oversampling

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means the change in sampling rate employed in today’s modern D/A converter chips themselves.

But let’s start at the beginning. What is the sampling frequency? For any digital storage or transmission it is necessary to have time discrete samples of the signal which has to be processed. I.e. the analog signal has to be sampled at discrete time intervals and later converted to digital numbers (also see section

Jitter Suppression and Clocking, p. 3). This sampling 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 domain, one has to sample 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 components are generated which are perceived as a very nasty kind of distortion. So if one deﬁnes the audio band of interest to lie between 20 Hz and 20 kHz, then the minimum sampling frequency for such signals must be 40 kHz.

For practical reasons explained below, the sampling frequency of 44.1 kHz

was chosen for the CD. A sampling frequency of 44.1 kHz allows to represent

signals up to 22.05 kHz. The designer of the system has to take care that any frequencies above 22.05 kHz are suﬃciently suppressed before sampling at

44.1 kHz. This suppression is done with the help of a low pass ﬁlter, which cuts oﬀ the frequencies above 22.05 kHz. In practice such a ﬁlter has a limited steepness, i.e. if it suppresses frequencies above 22.05 kHz it also suppresses frequencies between 20 kHz and 22.05 kHz to some extent. So in order to have a ﬁlter which suﬃciently suppresses frequencies above 22.05 kHz, one has to allow it to have a so-called transition band between 20 kHz and 22.05 kHz 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 more interest to the High-End Hi-Fi enthusiast, essentially the same ﬁlter is required. This is because after the D/A conversion we have a time discrete analog signal, i.e. a signal that looks like steps, having the rate of the sampling frequency.

Such a signal contains not only the original audio signal between 20 Hz and 20 kHz but also replicas of the same signal symmetrical around multiples of the sampling frequency. This may sound complicated, but the essence is that there are now signals above 22.05 kHz. These signals come from the sampling process.

There are now frequencies above 22.05 kHz which have to be suppressed, so that

they do not cause any intermodulation distortion in the ampliﬁers and speakers, do not burn tweeters or do not make the dog go mad.

Again, a low pass ﬁlter, which is called a”reconstruction ﬁlter“, is here to suppress those frequencies. The same applies to the reconstruction ﬁlter as to the anti-aliasing ﬁlter: pass-band up to 20 kHz, transition-band between 20 kHz and

22.05 kHz, stop-band above 22.05 kHz. You may think that such a ﬁlter is rather

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,,steep“, e.g. frequencies up to 20 kHz go through unaﬀected and frequencies above 22.05 kHz are suppressed to maybe

100 000

of their initial value. You are right, such a ﬁlter is very steep and as such has some nasty side eﬀects. For instance it does strange things to the phase near the cutoﬀ frequency (20 kHz) or it shows ringing due to the high steepness. In the early days of digital audio these side eﬀects have been recognized as being one of the main culprits for digital audio to sound bad.

So engineers looked for ways to enhance those ﬁlters. They can’t be eliminated because we are talking laws of physics here. But what if we run the whole thing at higher sampling rates? Like 96 kHz or so? With 96 kHz we can allow frequencies up to 48 kHz, so the reconstruction ﬁlter can have a transition band between

20 kHz and 48 kHz, a very much relaxed frequency response indeed. So let’s run

the whole at 96 kHz or even higher! Well — the CD stays at 44.1 kHz. So in order to have that analog lowpass ﬁlter (the reconstruction ﬁlter) to run at a relaxed frequency response we have to change the sampling frequency before the D/A process. Here is where the upsampler comes in. It takes the 44.1 kHz from the CD and upsamples it to 88.2 kHz or 176.4 kHz or even higher. The output of the upsampler is then fed to the D/A converters, which in turn feeds the reconstruction ﬁlter. All modern audio D/A converter chips have such an upsampler (or oversampler) already built into the chip. One particular chip, for instance, upsamples the signal by a factor of eight, i.e. 44.1 kHz ends up at

352.8 kHz. Such a high sampling frequency relaxes the job of the reconstruction ﬁlter very much; it can be built with a simple 3rdorder ﬁlter.

So, how come that upsamplers are such a big thing in High-End Hi-Fi circles?

The problem with the upsamplers is that they are ﬁlters again, digital ones,

but still ﬁlters. So in essence the problem of the analog reconstruction ﬁlter has been transferred to the digital domain into the upsampler ﬁlters. The big advantage when doing it in the digital domain is that it can be done with a linear phase response, which means that there are no strange phase shifts near

20 kHz and the ringing can also be controlled to some extent. Digital ﬁlters in

turn have other problems and of course have quite a few degrees of freedom for the designer to specify. This means that the quality of digital ﬁlters can vary at least as much as the quality of analog ﬁlters can. So for a High-End Hi-Fi designer it is a question whether the oversampling ﬁlter built into the D/A chips lives up to his/her expectations. If not, he/she can choose to design his/her own upsampler and bypass part of or the whole oversampler in the D/A chip. This gives the High-End Hi-Fi designer yet another degree of freedom to optimize the sonic quality of the product.

1.3 Dithering

You have probably not heard the term dithering in conjunction with audio. Actually it is a term widely used in the professional audio realm but not so much

in the High-End Hi-Fi market.

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What is dithering? Suppose a digital recording has been made 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 sample, as you know. What to do with those 8 bits, which are too many? The simplest way is to cut them oﬀ, truncate them. This, unfortunately, generates harmonic distortions at low levels, but

which nonetheless cause the audio to sound harsh and unpleasant. The harmonic

distortion is generated because the eight bits, which are cut oﬀ from 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 from 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

components but a slightly increased noise ﬂoor. This works like magic — the

distortion is replaced by a small noise which is much more pleasant. I have given

the example 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 more signal processing is

implemented in the digital domain. Think of digital equalizers, digital volume

controls, upsamplers, digital pre-ampliﬁers, decoders for encoded signals on DVD

etc. All those applications perform some mathematical operations on the digital

audio signal. This in turn causes the wordlength of the signal to be increased;

an input signal to an upsampler may have a wordlength of 16 bits (oﬀ a CD),

but the output signal of the upsampler may have 24 bits or even more. This

comes from the fact that the mathematical operations employed in such devices

increase the word length. A multiplication of two 2-digit numbers results in

a four-digit number. So after the upsampler the word length may be higher

than the subsequent processor may be able to accept. In this example, after

the upsampler there may be a D/A converter with a 24 bit input word length

capability. So if the upsampler generates a word length of more than 24 bits it

should be dithered to 24 bits for maximum signal ﬁdelity.

Another application where dither is important is described in the next

paragraph.

1.4 Digital Level Control

In High-End Hi-Fi circles a level control done in the digital domain is often

viewed as being inferior to one operating in the analog domain. Let’s look on

how a digital level control works and why it can be an excellent solution if it is

properly implemented.

A level control is a multiplication of the audio signal with a constant, the ,,gain factor“. The gain factor usually is in the range of zero (signal fully oﬀ ) to one (signal untouched). A factor of 0.5 then means that the audio signal is attenuated to half of its amplitude. What exactly happens when we multiply two numbers? If we e.g. multiply a 2 digit and a 3 digit number, the resulting

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number can be up to 5 digits long (the sum 2 plus 3). As an example: 30 times 500 equals 15 000; 2 digits times 3 digits yields a 5 digit result.

In digital audio, the numbers are represented in the binary system, not the decimal system. A decimal number consists of digits 0 through 9, a binary number of digits 0 and 1. So a binary number may look like this: 1011 0011 0101 1101.

This is a 16 digit or 16 bit binary number, the grouping into 4 bit chunks is

for better readability. The audio samples on a CD are represented with such a binary number system with each sample value represented with 16 bits. Now let’s assume we have a 8 bit gain factor for a level control. If we apply that to a signal coming oﬀ a CD we multiply a 8 bit gain factor with a 16 bit sample

value. The result is up to 24 bits long (the sum of the wordlengths of the two

factors). An example:

0100 1001 × 1001 0110 0111 1011 = 0010 1010 1110 1001 0001 0011

The question now is what do we do with the 24 bit long result? The digital to

analog converter which converts the samples after the level control may only be capable to handle 16 bit wide samples. Thus what should we do with the excessive 8 bits? The simplest solution is to truncate the 24 bit sample to 16 bits, i.e. to cut oﬀ the 8 least signiﬁcant bits. The truncated 24 bit result above then

would look like this: 0010 1010 1110 1001 i.e. the ﬁrst 16 bits of the 24 bit result

above.

The remaining bits (0001 0011) are discarded. If these bits are discarded an error is introduced. This error is called a quantization error, because the 24 bit result is requantized to 16 bits. Unfortunately the quantization error is part of the audio signal — and if we take that part away from the signal, the signal undergoes some distortion, the so called quantization distortion. The sound example at the link below shows how such a distortion sounds. In this music example a 16 bit signal is truncated to 8 bits; 8 bits in order to clearly show the eﬀect. Notice how the noise (distortion) is modulated by the music signal.

www.weiss.ch/linked/digital-level-control/nodither.mp3

This is how a badly implemented digital level control works. . . Fortunately there

is a better way to handle the re-quantizing. One solution would be to use a D/A converter with a higher wordlength, e.g. a 24 bit converter, to accommodate for the 24 bit samples coming out of the level control. This of course would already help a lot, but there is another technique: dithering.

The idea about dithering is to de-correlate the quantization error from the audio signal. As we have seen in the example above, the quantization error depends on the audio signal (i.e. it is correlated with the audio signal). On the other hand, if dither noise is added to the 24 bit sample after the level control and before the re-quantization to 16 bits, the quantization error can be fully decorrelated from the signal. This means instead of distortion there is noise.

The music is undistorted. The audio example at the link below is again a 16 bit

signal quantized to 8 bits, but with dither noise added. A much more pleasant

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experience. Notice how the noise stays untouched by the music, i.e. there is no noise modulation.

www.weiss.ch/linked/digital-level-control/ﬂatdither.mp3

Dithering does not stop here. More elaborate dithering schemes shape the noise such that it is mainly present at higher frequencies where the human ear is less sensitive. This means that the audible noise is much lower. The link below is again the 16 bit source quantized to 8 bits with noise-shaped dithering. Probably hard to believe that this is only an 8 bit system! Note that the music is not distorted at all, despite the 8 bit resolution. Remember, a 16 bit system has 65 536 quantization steps while a 8 bit system has only 256 quantization steps — a huge diﬀerence. And still, the properly dithered 8 bit system sounds great.

www.weiss.ch/linked/digital-level-control/shapeddither.mp3

This is what a properly dithered level control is capable to do. You have heard

the 8 bit version, imagine that with today’s 24 bit converters — no question that a level control with a 24 bit wordlength easily rivals the best analog level controls. By the way, 24 bits means 16 777 216 quantization steps.

The last example below toggles the noiseshaping dither on and oﬀ to give a

good contrast between dither/no dither versions.

www.weiss.ch/linked/digital-level-control/togglingdither.mp3

Dithering is used in many disciplines. Also see

en.wikipedia.org/wiki/Dithering

A paper on this topic (including image examples) can be found here:

www.weiss.ch/assets/content/41/white-paper-on-digital-level-control.pdf

I hope these excursions into the theory and practice of audio engineering has been useful for you. If you would like to dive further into those issues I recommend visiting the website of Mr. Bob Katz, a renowned mastering engineer and a Weiss Engineering ltd . customer. He publishes articles on dithering and jitter and many other topics at

www.digido.com

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Chapter 2

The INT204 USB/DSD

Interface

2.1 Features

2.1.1 Input

There is one USB type B socket for connecting a computer. The supported

sampling frequencies are 44.1, 48.0, 88.2, 96.0, 176.4, 192.0, 352.8, 384.0 kHz and DSD over PCM (DoP) at 2.8224 MHz and 5.6448 MHz (DSD64, DSD128).

2.1.2 Outputs

There is one each of XLR, BNC and RCA connectors available on the back

panel. The XLR has its own driver while the RCA and BNC connectors have one common driver. All three outputs carry the same data.

2.1.3 Synchronization

The synchronization master is always the INT204 device, i.e. the computer is

slaved to the INT204. The sampling frequency is set via the USB driver. On a Mac system the sampling frequency is set in AudioMidi application. With the proper (advanced) player software the sampling frequency of the INT204 is switched automatically depending on the track played. The INT204’s intrinsic jitter is very low.

2.1.4 DSD to PCM conversion

In case of DSD playback, the DSD signal is converted to PCM. The sampling frequency and word length of the PCM signal can be chosen to match the capabilities of the D/A Converter connected to the INT204.

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2.1.5 Power Supply

The INT204 uses two power sources. One is the bus power coming via the USB

connector. It powers the USB input section of the INT204, which is isolated from the rest of the circuit. The rest of the circuit (output section etc.) is powered

via the external power supply which is supplied with the INT204.

2.1.6 Front Panel

• Power switch

•

DSD conversion mode switches for word length and sampling frequency selection

• LEDs for power indication and sampling frequency indication

• IR Receiver

2.1.7 Back Panel

• (1) XLR connector

• (1) RCA connector

• (1) BNC connector

• (1) USB type B connector

• (1) Power connector

2.2 Installation

Carefully unpack the INT204. The following items should be enclosed:

• the INT204 USB/DSD Interface unit

• a Power Supply

•

a CD with the necessary drivers for Windows. OSX does not require any driver.

• this Owner’s Manual

• a Certiﬁcate of Guarantee

2.2.1 Software installation

Perform the following installation procedure before connecting the INT204 to the computer. The necessary ﬁles are supplied on the enclosed driver CD.

For Windows:

• Launch SETUP.EXE found on the driver CD and follow the instructions.

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Figure 2.1: INT204 frontpanel

Figure 2.2: INT204 backpanel

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•

Click on

when MS Windows warns you about unveriﬁed driver

installation.

• Plug the USB cable according the installation wizard messages.

• Use MS Windows control panel to set up the audio interface correctly.

• Conﬁgure your media player to use the USB Audio ASIO driver.

For OSX:

• No driver to be installed.

2.3 Operation

2.3.1 USB and Power Connection

First install the software on your computer as described in section 2.2.1 (not necessary for OSX). Connect the computer to the INT204 via an appropriate USB cable. Use an USB cable as short as practicable. Connect the enclosed power supply to the power inlet at the INT204. Switch on the computer and the INT204. The INT204 should now be recognized by the computer and should appear in the audio devices settings. Select the INT204 as the playback device.

2.3.2 AES/EBU or S/PDIF Connection

Connect the output of the INT204 to your D/A Converter or any other digital audio device. Use either the XLR or RCA or BNC connectors as applicable.

You may connect more than one D/A Converter simultaneously. When using

the BNC and RCA outputs simultaneously it may fail to work though, as the BNC and RCA connectors are driven by a single driver. But do not hesitate to try. Make sure to use an appropriate cable to connect the D/A Converter to the INT204. E.g. the XLR cable must be one suited for AES/EBU signals. An

”

analog“ type AES/EBU cable must not be used.

2.3.3 DSD to PCM Conversion

•

Sampling frequency: 176.4 or 88.2 kHz. Use the 88.2 kHz setting if your D/A Converter cannot handle the 176.4 kHz sampling frequency.

•

Word length: 24 or 16 bit. Use the 16 bit setting if your D/A Converter cannot handle a word length of 24 bit.

The conversion of the DSD signal to PCM before the D/A conversion has some

advantages compared to a native DSD D/A conversion. The white paper in the appendix A gives details about that topic.

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2.3.4 PCM to PCM Conversion

In case of 352.8 kHz playback, the sampling frequency will be converted to 176.4 kHz. In case of 384.0 kHz playback, the sampling frequency will be converted to

192.0 kHz.

2.3.5 LED Sampling Frequency Indication

The 8 LEDs indicate the power status as well as the input sampling frequency

and format. In case of 352.8 kHz or 384.0 kHz input sampling frequencies only the power LED is lit.

2.3.6 Optional IR Level Control

There is an optional IR Remote Control unit available for the INT204. This

allows to control the volume in the digital domain. If the IR Remote Control is not included with the INT204, the volume control is factory set to 0 dB gain, i.e. there isn’t any inﬂuence on the audio signal.

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Figure 2.3: INT204 remote control

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Chapter 3

INT204 Technical Data

3.1 Digital Inputs

• (1) USB type B connector

•

Sampling Frequencies: 44.1, 48.0, 88.2, 96.0, 176.4, 192.0, 352.8, 384.0 kHz or DSD over PCM (DoP) at 2.8224 or 5.6448 MHz.

• Maximum input wordlength: 24 bits

3.2 Digital Outputs

• (1) XLR connector

• (1) RCA connector

• (1) BNC connector

• Channel Status Data bits are set to all 0 on all three outputs

3.3 Synchronization

•

Synchronized via the clock internal to the INT204, i.e. the computer is slaved to the INT204

•

Sampling Frequencies: 44.1, 48.0, 88.2, 96.0, 176.4, 192.0, 352.8 or 384.0kHz

3.4 DSD Conversion

The DSD to PCM conversion can be conﬁgured to match the capabilities of the

D/A Converter connected to the INT204. The sampling frequency can be chosen between 176.4 and 88.2 kHz while the output word length can be chosen between 24 and 16 bit.

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3.5 Power

Bus power via the USB input for the USB input section. External power supply for the output section. External power supply voltage: 6–9 V DC at 0.15 A.

The USB input is isolated from the digital audio outputs and from the external

power supply.

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Appendix A

Some information on the DSD format — properties, caveats, uses.

The DSD (Direct Stream Digital) Audio format is used with the well-known

SACD disc. The SACD has some form of copy protection implemented, thus an SACD is not easily rip-able into a computer ﬁle.

A.1 Quantizing

DSD is a digital signal which uses a 1 bit quantization of the audio signal at a sampling rate of 64 times the sampling rate of a CD (64×44100Hz= 2.8224 MHz).

The CD format uses a 16 bit quantization at a 44.1 kHz sampling rate, thus the

data rate of a DSD signal is four times the data rate of a CD, which gives the SACD the potential to represent the audio signal more precisely. The noise ﬂoor of a signal at 44.1 kHz/16 bit is about 16×6dB= 96 dB below full scale. With the DSD signal the noise ﬂoor is 1×6 dB below full scale, so obviously there have

to be taken some measures to get down the noise ﬂoor, else a DSD recording would be very noisy. Here the high sampling rate of DSD comes in handy. I.e. with special ﬁltering techniques (noise-shaping) it is possible to move the noise

in the audio band (0–20 kHz) out to higher frequencies above 20 kHz. Thus the

noise ﬂoor in the audio band can be brought down to very low levels (well below

the 16 bit noise ﬂoor of a CD) at the cost that the noise is at a fairly high level

at frequencies above 20 kHz.

A.2 Filtering

This high level/high frequency noise has to be suppressed before the signal

hits the ampliﬁer and speakers, in order to avoid ampliﬁer misbehaving and/or

speaker blow-up. A low pass ﬁlter is used to suppress the noise. In a native

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DSD D/A Converter this low pass ﬁlter has to be implemented in the analog domain which can be problematic because the non-linear phase characteristics of such a ﬁlter. Alternatively this ﬁlter can be implemented in the digital domain

where the DSD signal is converted to PCM (at e.g. 176.4 kHz/24 bit) and at the

same time gets properly low pass ﬁltered, e.g. with a linear phase ﬁlter which is easily achievable in the digital domain.

A.3 DSD uses

The DSD format initially was conceived for distribution and archiving. Later on

it became more and more popular in music recording with its own set of problems as described below. DSD shows the state of the art of the 90ies. The 1 bit sigma-delta A/D and D/A converters were very popular at that time. Converter techniques have progressed though. Today the modern sigma-delta converters are multi-bit designs with e.g. 6 bits instead of 1 bit. This for good reasons of course, the 1 bit technique has some inherent problems with dithering and idle tones. If the SACD was conceived today, it would use a 6 bit format.

A.4 Processing

If a recording made in DSD has to be processed, e.g. the loudness has to be adjusted or some equalizing is required, then the DSD signal has to be converted to PCM in order to be able to do the processing. In the PCM format the signal can be processed, i.e.”produced“ as required in order to get a ﬁnal version. In such a case it does not make sense to do a recording in DSD to start with. A PCM recording at e.g. 88.2 kHz/24bit would make much more sense. After the production process the ﬁle can always be converted to DSD if required.

A.5 What is better?

The discussion whether DSD is better than PCM is moot. DSD is another variant to store music, like PCM, MP3, magnetic tape, vinyl etc. All formats

are ﬁne as long as there are people preferring one over the other for whatever reasons. A matter of taste or practical reasons as with MP3 for instance.

A.6 The future

Technically speaking, PCM from like 88.2 kHz/24 bit up, is deﬁnitively better

than DSD. There are attempts to enhance DSD by rising the sampling rate from

2.8824 MHz to 5.6448 MHz or even 11.2896 MHz. Some problems of DSD may be reduced with such measures, but the 1 bit quantization problem cannot be eliminated. With the DSD downloads becoming more and more popular the DSD format is going to stay with us. A niche, but a growing one.

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Appendix B

Contact

For any questions, suggestions etc. feel free to contact us at:

Weiss Engineering ltd . Florastrasse 42 8610 Uster Switzerland

Phone: +41 44 940 20 06 Fax: +41 44 940 22 14 Email: [email protected] Web: www.weiss.ch

www.asiaweiss.com

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