Audio Critic the 21 r schematic

Issue No. 21

Display until arrival of

Issue No. 22.

Retail price: U.S. $7.50, Can. $8.50

Unconventional deployment of conventional drivers,
with superior results. (See the loudspeaker reviews.)

In this issue:

We present the definitive article for audiophiles on
the subject of digital jitter, written by one of the
world's top experts in response to the appalling
misinformation spread by the high-end audio press.

Your Editor reviews an unusually interesting and
varied assortment of loudspeaker systems.

We take a first look at the Sony MiniDisc system.
David Rich dissects analog and digital electronics

by Harman Kardon, Krell, Parasound, Sentec, et al.
Plus many other test reports, all our

regular columns, letters to the Editor,
and CD reviews by David Ranada.

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Contents

10

Clock Jitter, D/A Converters, and Sample-Rate Conversion

By Robert W. Adams, Analog Devices, Inc., Wilmington, MA

25

Loudspeaker Systems Using Forward-Firing Cones and
Domes: How Good Can They Get?

By Peter Aczel, Editor and Publisher

25 B&W Matrix 803 Series 2
26 NHT Model 3.3
28 Sequerra Model NFM-PRO
29 Velodyne DF-661 (quick preview)
30 Magneplanar MG-1.5/QR (dipole)
31 Bag End ELF Systems S10E-C and S18E-C (subwoofers)
33 MSB Acoustic Screens (acoustical accessory)

34

Analog Electronics: More Power Amplifiers,
Preamp/Control Units, and Mild Surprises

By David A. Rich, Ph.D., Contributing Technical Editor

34 Stereo Power Amplifier: Harman Kardon PA2400
36 Full-Function Preamplifier: Harman Kardon AP2500

38 Line-Level Remote Preamplifier: Krell KRC-2
41 Stereo Power Amplifier: Parasound HCA-2200II
44 Line-Level Preamplifier: Rotel RHA-10 (quick preview by the Editor)
44 Stereo Power Amplifier: Rotel RHB-10 (quick preview by the Editor)

45

Digital Electronics: More CD Players, D/A Processors,
Transports, and a First Look at the Sony MiniDisc System

By Peter Aczel, Editor and Publisher
& David A. Rich, Ph.D., Contributing Technical Editor

45 Outboard D/A Converter: EAD DSP-90000 Pro (Reviewed by Peter Aczel)
46 Compact Disc Player: Harman Kardon HD7725 (Reviewed by David Rich)
48 Outboard D/A Converter: Krell Studio (Reviewed by David Rich)
50 DAC/Line Amplifier: Monarchy Audio Model 33 (Reviewed by Peter Aczel)
53 CD/Videodisc Transport: Monarchy Audio DT-40A (Reviewed by Peter Aczel)
53 Outboard D/A Converter: Sentec DiAna (Reviewed by David Rich)
55 Compact Disc Player: Sony CDP-X707ES (Reviewed by Peter Aczel)
56 2nd-Generation MiniDisc Recorder: Sony MDS-501 (Reviewed by Peter Aczel)

57

High-Definition Thinking in Small, Furry Mammals

or The Weasel's Guide to Maximum Satisfaction

By Tom Nousaine

60

Hip Boots Wading through the Mire of Misinformation in the Audio Press

Six commentaries by the Editor

62

Recorded Music

A

Miscellany of CDs and Musical Videodiscs

By David Ranada (with a few additional capsule reviews by the Editor)

3 Box 978: Letters to the Editor

ISSUE NO. 21 • SPRING 1994 1

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Issue No. 21 Spring 1994
Editor and Publisher Peter Aczel

Contributing Technical Editor David Rich
Contributing Editor at Large David Ranada
Technical Consultant Steven Norsworthy
Columnist Tom Nousaine
Cartoonist and Illustrator Tom Aczel
Business Manager Bodil Aczel

The Audio Critic® (ISSN 0146-4701) is published quarterly for $24

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Send address changes to The Audio Critic, P.O. Box 978, Quakertown,
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The Audio Critic is an advisory service and technical review for
consumers of sophisticated audio equipment. Any conclusion, rating,
recommendation, criticism, or caveat published by The Audio Critic
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Contents of this issue copyright © 1994 by Critic Publications, Inc. All rights
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From the Editor/Publisher:

Issue No. 20 was optimistically dated Late
Summer 1993 but was mailed in the second
week of the fall. Unforeseen delays resulted
in one omitted quarter; hence the more
realistic Spring 1994 dating of this issue.
The staff expansion I so fondly previewed
is still in the incipient stage. With all the
reviews in this issue, it was suggested to
me that I split it down the middle to make it
into two issues (maybe I should have), or
label it a double issue in fulfillment of two
quarters of a subscription (I would never
do that). Yes, there will be a Summer 1994
issue, possibly even sooner than you think.

2 THE AUDIO CRITIC

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Box 978

Letters to the Editor

In the last issue your Ed. griped in this space about uninteresting letters seeking advice on purely

private purchasing plans (alliteration unintended). Now I want to gripe about reasonably interesting

but unpublishable letters that ramble on for seven or eight illegibly handwritten pages, propounding

the correspondent's opinions on eleven different audio subjects. What is the purpose of such a letter?

What am I supposed to do with it? Get a life, guys—or get a word processor. Letters printed here
may or may not be excerpted at the discretion of the Editor. Ellipsis (...) indicates omission. Address
all editorial correspondence to the Editor, The Audio Critic, P.O. Box 978, Quakertown, PA 18951.

The Audio Critic:

When making the suggestion that
Berlioz was worthy of joining "the 3 B's"
[Issue No. 17, p. 57], you likely never
knew that Liszt's disciple Peter Cornelius
coined the formula with Berlioz as the
third! Bülow signaled his revolt from
Wagner (who had earlier stolen his wife
Cosima, née Liszt) by purloining the
phrase to glorify Brahms.

Actually "the 3 B's" may be held
nonsense by Magyars, or other non­Teutons (such as I). So is your warning
that Berlioz's worst is probably worse
than Bach's, Beethoven's, or Brahms's
worst. Since Berlioz is the lone great
composer who was never granted full
"canonization" (which can account for
your electing him so shyly to "3 B" status,
and the rare event of Inbal's semicycle
recordings: he'll never be subjected to the
fashion), he still suffers from such pre­posterous slights, even from renowned
musicians (e.g., Celibidache and Nigel
Kennedy), which only an exceedingly bold
critic would make against the others.

Can you recall Tovey's remarks—of
which Haggin was so fond—that "neither
Shakespeare nor Schubert will ever be
understood by any critic or artist who re-

gards their weaknesses and inequalities
as proof that they are artists of less than
the highest rank," and that "the highest
qualities attained in important [my em­phasis] parts of a great work are as inde­structible by weaknesses elsewhere as if
the weaknesses were the accidents of
physical ruin." Tovey wrote that to exalt
Schubert against the shallow regard then

rampant but now all but extinct....

I don't overlook the fact that the mu­sic of Berlioz pleases you greatly. (A vis­itor to Wagner reported that he first said
that Romeo and Juliet's Love Scene, just
as wisely called the Adagio of Berlioz's
3rd Symphony, was "the most beautiful
music ever composed.") A superior critic
(and Nobel-winning novelist) of France,
Romain Rolland (also friend and corre­spondent of Freud!) enrolled Bach, Han­del, Mozart, Beethoven, Schubert, and
Wagner as the finest of all composers and
claimed that after them he knew no other
who was superior, even equal, to Hector
Berlioz. Berlioz seemed incapable of the
vulgarity (or kitsch or bathos) which
afflicts German composers, and Brahms
especially (ever seen Cary Grant conduct
the Academic Festival Overture in Peo-

ple Will Talk?), and so became approved

and emulated in all Western music...

...While comparing one and an­other's worst and best, what is gained
from tallying pages, measuring the stacks
of bad and good? Yet you felt compelled
to indulge this pointless fancy at Berlioz's

expense....

Sincerely,
Owen M. Feldman
Elkins Park, PA

Ha! Fooled you all—didn't I?—by
beginning this column with a music­oriented letter rather than audio talk. My
main reason for doing so was actually to
contradict my preamble above with, shall
we say, the exception that proves the
rule. This letter came written in a small,
compressed hand on what appears to be
a legal-size yellow pad, then very badly
Xeroxed and the copy sent instead of the
original. I chewed my way through 3¼
legal-size pages of smeared, streaked,
gray mess and decided to publish about a

fourth of it because I found it interesting

and entertaining. I'm not taking back a
single word of my preambulary com­ments, but I guess I am a sucker for
knowledgeable music talk. It's certainly a
nice change from tweako audio talk. I'll

ISSUE NO. 21 • SPRING 1994 3

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even apologize for my churlishness anent
Berlioz's small lapses; on second thought
they're probably no worse than Bach's or
Beethoven's. Besides, a man who is even
slightly underwhelmed by Brahms can't
be all bad and should be humored.

-Ed.

The Audio Critic:

..."Accountability in audio journal­ism" and your no-nonsense, rational ap­proach to product reviews are a refresh­ing change and a source of continuing
entertainment for me. David Rich is a su­perb find.

Regarding the MTM [mid/tweet/mid]
driver geometry, you are correct in that I
was not the first to use it (Issue No. 20,
page 42). To my knowledge the earliest
commercially successful use was by Koss
in a small 2-way system with 4" mid/bass
drivers. The choice of this geometry by
Koss, however, appeared to be largely
cosmetic. Meridian also made such a sys­tem about the time my paper appeared
["A Geometric Approach to Eliminating
Lobing Error in Multiway Loudspeakers,"
74th Convention of the AES, New York,
8-12 October 1983, Preprint 2000], but
again no mention of its superior polar re­sponse was made by them.

With Linkwitz's 1976 paper the
problem of polar-axis frequency­dependent wander or lobing error became
widely appreciated. Linkwitz's solution
to the problem was to use inphase cross­over networks. I was the first to demon­strate in the open literature that the MTM
geometry automatically eliminates lobing
error and to show the relationship be­tween polar response and crossover order
for this geometry. I also designed several
commercially successful loudspeaker sys­tems and system kits using this geometry.

Two of these systems were featured in
Speaker Builder magazine. The "Auditor
Point Source Aria Five," a Focal/JML
product which sold exlusively in Europe,
won the best loudspeaker of the year
award for 1991 from Hifi Vidéo (Paris,
March 1991). The MTM geometry is
now widely used and several manufactur­ers have attributed the concept to me in
their promotional literature. I believe it is
for these reasons that the MTM geometry
has become associated with my name.

Yours truly,

Joseph D'Appolito, Ph.D.

Andover, MA

Thank you for the compliments. Isn't

it remarkable that technologists with the
highest credentials, such as you, always
like us and that the scattered little en­claves of hostility out there are invariably

peopled by the technically untutored?

As for the MTM geometry, I myself
was the grunt of a design team (Bruce
Zayde, now with Hewlett-Packard, was
the whiz) that developed such a speaker
in 1984-85. It was called the Fourier 44
(because of the two 4½" mid/bass driv­ers) and shown at the 1985 Summer CES.
A few studio types and broadcasters are
still using it as a small monitor. We
didn't attach the D'Appolito appellation
to it, but conceptually the crossover was
along the Linkwitz/D'Appolito guidelines.
The speaker is currently extinct.

Audio designers have been known to
claim credit for work done by others, but
you are the first in my experience to dis­claim, or heavily circumscribe, credit for
something the world has already fully
credited to you. That's what I call a class
act!

—Ed.

The Audio Critic:

Sorry, Charlie! This $24 is going to
Stereophilel I am sitting in the smallest
room of my house with your so-called
magazine in front of me. It will soon be

behind me. I'm sorry I ever wasted a cent
on your rag.

[—Unsigned]

The above anonymous and untrace-

able message was scribbled on a blank
copy of our pink form soliciting renewal
of an expired subscription and returned
to us in our business reply envelope at
our expense. I am publishing it as a clue
to the sociocultural/intellectual profile of
those who opt for Stereophile in prefer­ence to our publication. Such class! Such
wit! But such an awkward seat for letter
writing! (Needless to say, I washed my
hands after handling the form.)

—Ed.

The Audio Critic:

In Issue No. 20, Drew Daniels' letter
cited a transmission line's electrical
length as analogous to the phase shift in a
length of speaker cable. This is not true.
A speaker cable is in effect a lowpass
filter—your own curves in Issue No. 16
clearly show this. Depending on the LCR
of the cable, the driving impedance (am­plifier), and the load impedance (speak­er), the cutoff frequency could take place

within or, hopefully, above the audio
band.

The phase shift within the audio
band is determined by all of the above,
but generally will be capacitive (negative
degrees) at low frequencies, pass through
resistive (zero degrees), then become in-

ductive (positive degrees) at higher fre­quencies. In any case, even modest
lengths of any of the cables sold today
will exibit many degrees of phase shift at
the speaker terminals as a function of fre­quency. The only way to avoid power
loss, high-frequency rolloff, reduced
damping factor (drastic in some cases),
and phase shift (although I don't know
that this is important) is to use no speaker
cable. Your own suggestion to use mono
amps at the speakers' backs with short

jumpers is a most valid one.

Another subject: May I respectfully

decline to accept your request to contrib-

ute for the advancement of Bob Harley's

technical education? I find his "jittery"
stepping through technical issues very

amusing and entertaining. With more ed­ucation he could become dangerous—
another Martin Colloms!

Don't give in to those who would
like you to compromise with the witch
doctors. Someone has to bring all the
hype to the forefront and it appears you
are the only one.

Sincerely,
Jefferson P. Lamb

Incline Village, NV

It seems to me that Drew Daniels
and you are talking about two different
things. He talks about phase shift as re­lated to propagation speed in an untermi­nated wire, which is an abstraction; you
talk about a real-world hookup with an
amplifier output impedance plus wire
characteristics plus a complex load impe­dance presented by the speaker. Yes, of
course, with your givens your conclu­sions apply, but then the issue becomes
the sound of the amplifier/wire/speaker,
not just the sound of the wire due to its
length, which is the "moronic" subject
that incenses Drew.

Anent the SHEESH (Send Harley to

E.E. School in a Hurry) Fund, see the

"Hip Boots" column in this issue. Now,
what you don't seem to realize is that
Martin Colloms is actually two persons
existing in parallel universes. There is
the techie Martin Colloms, made of mat­ter. There is the tweako cultist Martin
Colloms, made of antimatter. The two

4 THE AUDIO CRITIC

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cannot possibly get together because that
would cause a cataclysmic explosion an­nihilating Hi-Fi News & Record Review
and maybe even Stereophile. That's why

he is dangerous.

—Ed.

The Audio Critic:

"Dear" Peter,

Your continued hysterical, personal
attacks on me are entertaining as usual.
As is Ken Pohlmann's letter, which ex­plains his affinity for you: he too is a pet­ulant, name-calling infant. Why don't
you do this: survey the top LP and CD
masterers in the field—the folks who
have access to the master tapes and the
CD transfers. Ask them, as I have,
whether the CDs—especially those made
from an analogue tape—sound like the
original. And ask them whether they pre­fer the sound of a CD or a properly man­ufactured and played-back LP.

Ask Doug Sax, Bernie Grundman,
Greg Calbi, Bob Ludwig, Ted Jensen,
Stephen Marcussen, Steve Hoffman,
Howie Weinberg, Bill Inglot, etc. Ask
Grammy-award-winning engineer Roy
Halee (Paul Simon, etc.) about digital re­cording, or any of the dozens of veteran
recording engineers I've surveyed on the
subject. Some like digital, many don't.
I'd be happy to provide you with a list. [I
thought you just did.—Ed..]

Here's a good one: call veteran jazz
producer Michael Cuscuna of Mosaic
Records and ask him what happened
when, without alerting his customers, he
started releasing LPs generated from
"perfect" digitally remastered analog
source material. He'll tell you this: virtu­ally all of them heard the deleterious ef­fects of digitization and called to com-

plain.

Or better yet, reprint the following
so all five of your new readers [currently

a minimum of five new readers a day, Mi-

chael, but usually more—Ed.] can read

it:

Help Stop the Digital Epidemic!

It has become a mindlessly parrot­ed truism in the world of commer­cial audio that digital recording is
the state of the art and the wave of
the future. At the same time, there
isn't a single audiophile-oriented
equipment reviewer, record pro­ducer or music critic who finds the
treble range of current digital re­cordings musically natural and en­joyable. The present technology of
50,000 samples per second with

16-bit encoding/decoding is sim-

ISSUE NO.21 • SPRING 1994

ply inadequate and mustn't be al­lowed to become the world stan­dard. If you agree with us, start
writing letters to the record compa­nies and commercial magazines
before it's too late.

Let's see, who wrote that? Could it
be...SATAN? NO. It's from The Audio
Critic, Spring through Fall 1980, and

judging by the tone I'd say you wrote it,

you opportunistic slug.

Cheers,
Michael Fremer
Senior Editor

The Absolute Sound

Sea Cliff, NY

If listing Senator Robert Dole as one
of the anti-Clinton Republicans consti­tutes a hysterical, personal attack on him,
then listing you as one of the antidigital
audio journalist—which was all I did (Is-

sue No. 20, p. 13)—constitutes a hysteri-

cal, personal attack on you. It seems to

me, however, that once again (see Issue
No. 14, pp. 9 and 51) you are the pot

calling the kettle black. Isn't calling Ken

Pohlmann "a petulant, name-calling in-

fant" an act of name-calling—both hys-

terical and personal? Weren't your re-

marks about him in the Winter 1993 issue
of The Absolute Sound, threatening to
vomit on him at a CES dinner, infantile,
personal, and name-calling? Let me do
some fact-calling: Ken Pohlmann has a
graduate degree in E.E.; you do not. Ken
Pohlmann is a professor at a major uni­versity; you are not. Ken Pohlmann is the
author of a basic textbook on digital
audio; you are not. Ken Pohlmann is a

Vice President of the Audio Engineering

Society; you are not. Shall I go on? Shall
I insult you with a spelled-out conclu­sion? I think that will be unnecessary.

As for your list of names—a couple
of them well-known, others less so—you
say "some like digital, many don't," so
what's your point? Which of them don't?
I can trump each digitophobic name you
bring up with three world-class names to
the contrary; it's a silly game. The point
is not who likes digital; the point is the
inherent accuracy, or lack thereof, of the
linear PCM technology and of A/D and
D/A conversion. That's an important sub-

ject that needs to be addressed repeatedly

and is the main reason I am answering
your trivial crank letter at such length.

No one claims, I least of all, that
everything that has ever appeared on CD
sounds great. There are plenty of oppor­tunities to mess up between the first A/D

and last D/A stage in the recording and
playback chain; it used to happen often
but now it's much rarer. The basic ques­tion, and the only intelligent one, is this:
between a state-of-the-art DDD compact
disc and a state-of-the-art all-analog vi­nyl disc, each free from all the possible
technical goofs, which reproduces with
greater accuracy and fewer spuriae the
signal from the microphones? If anybody
thinks the answer is uncertain (as it is
not), let me add one other reasonable
constraint: 50 playbacks. Have I made
my point? If you eliminate the vinyl, the
picture changes; today's best analog and
digital master tapes can sound quite com­parable overall, but the signal-processing
possibilities are much more limited with
analog, and as a storage format digital is
considerably more stable.

That brings me to my 1980 com­ments on "the digital epidemic." There
were no CDs in those days; the only
readily available examples of the new
digital audio technology were vinyl LPs
cut from digital master tapes. They
sounded pretty bad. I mistakenly attribut­ed the bad sound to the digital process. I
was dead wrong, as later CD releases of
those same digital master tapes clearly
showed. It was the transfer to the LP me­dium that was bad because the engineers
who did the transfers initially refused to
deal with the differences in high-frequency
energy and dynamic range between ana­log and digital master tapes. I had no
idea how the digital tapes sounded in the
control room at the recording sessions. I

jumped to the wrong conclusion, and my

advisors at the time were no smarter.

Have you ever jumped to the wrong con­clusion, Michael?

As for "opportunistic slug"—boy,
that's a muddleheaded, inept insult. What
opportunities does a snail without a shell
exploit unfairly? What undeserved advan­tages did I gain by repudiating and cor­recting my early perceptions of digital
audio? You, sir, are a very lightweight

polemist.

—Ed.

* * *

Epilogue: At the Winter CES in Las

Vegas early in January, Michael Fremer

deliberately staged a loud, public,

fishwifely confrontation with me for the

benefit of visitors to the Velodyne exhibit.
It was uncalled-for and embarrassing; I
later had to clear the air with David
Hall, Velodyne's boss, who is a soft­spoken gentleman unaccustomed to the

5

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streets-of-New-York type of vulgarity. I
tried to calm Michael by telling him that I
did not consider him to be an evil person
but just someone with absurd ideas, but
he kept loudly accusing me of "ad homi­nem attacks" (his editor, Harry Pearson,
also loves that phrase) and looking
around for group approval. I then made
the mistake of tossing off a small pedan­tic joke. I asked, "Who was the homo in

hominem?" Then I added, "You know,
hominem is the accusative of homo, re-

quired by the preposition ad." Michael
did not get the Latinist jest. His confused
reaction indicated that all he had heard
was the word homo, and I'm not even
sure he knew it means man. That will
teach me to make gratuitous scholarly
noises where the cultural tone is New
York candy store. Indeed, that will teach

me to have any kind of commerce with

the Fremeroid element in the audio world.

The Audio Critic:

It's been roughly two years or so

since my departure from "tweako/voodoo

salon" land, and I owe thanks to you and

your staff for the insight and knowledge
you share through The Audio Critic. It's
amazing what science and a little com­mon sense can do for the soul. As the
song states, "I was once blind but now I
see!" My only regret is that some of the
information is a little technical for those
"laymen" who are not a part of the engi­neering kingdom. I would be grateful if
you could dilute some of the techno-lingo
from time to time.

Throughout the years, I've noticed
that many audio magazines rarely even
mention the name McIntosh. The compa­ny has been around since 1949 and has a
solid reputation for reliability and quality,

just like Krell, Bryston, etc., yet the

tweaks hardly touch it. Why? The design
(external appearance) may be a bit out of

date for some, but the internal compo­nents hardly seem archaic. According to
the principles presented in your publica­tion, if the McIntosh amps operate within
their given parameters, then they should

sound no different than a Krell or Boul-

der. Thus, these amps should be highly
regarded and recommended, unless
there's something I've missed. How
about The Audio Critic, in a future issue,
taking the opportunity to test a McIntosh
amp. I would love to see how it compares
to some of the other big boys on the

block....

Please keep up the excellent work.

6

Your magazine lights the path for many

in a confused and disturbed audio world.

Sincerely,

Mark S. Williamson

Washington, D.C.

/ blush because your words of praise

make me appear to be something close to
a spiritual leader, a responsibility I re-

fuse to shoulder. (Lenny Bruce once said

that anyone who calls himself a religious
leader and owns more than one suit is a
hustler. I own two suits.)

McIntosh is indeed an interesting
company. You are quite right; they make,
and always made, beautifully engineered
equipment. During their long years under
the leadership of the late Gordon Gow,
they kept their dialogue with the high-end

audio press to a minimum, probably be-

cause they felt that with their thoroughly
established reputation and highly suppor­tive dealer network they had nothing to

gain from a good review but something to

lose as a result of an irresponsible tweako

hatchet job. Hey, they were probably

right.

About two years ago the picture
changed somewhat. The firm is now
owned by the Japanese; the new manage­ment is not from the high-end audio
world and is gung ho on marketing, PR,
the whole big-business canon. They have
a new car-audio line, among other
things. So far, from where I'm sitting, I
can discern no compromise whatsoever
with traditional Mcintosh engineering or

product integrity, and the party line is

that there will be none. Amen. They are
definitely cozier with the audio press,
however; you will undoubtedly see more
reviews, and I think that will include re­views in this publication. Based on what I

already know, I expect their power ampli-

fiers, especially, to do very well in engi-

neering shootouts with some of the sacred
cows of the High End.

—Ed.

The Audio Critic:

I just read my first issue of The Audio
Critic (No. 20). I can only say that I have

been looking for this type of coverage of

the audio industry for several years and
have found it only in this one publication.
I first became involved in audio as a teen­ager in 1966, working in what was then a
"high-end" audio store in San Francisco.
I have remained interested and involved
ever since.

I have been an avid reader of all the

popular audio publications over the years
except for Stereophile and The Absolute
Sound (by the time I discovered these last
two, they had become too involved in be­liefs in the superiority of older tube and
analog equipment for my taste). I have
had a subscription to Audio since the late
'60s. In recent years I have become high­ly disillusioned with the direction of most
publications and audio salons. So-called
"high-end" audio is becoming more and
more an exercise in frustration rather than
the source of pleasure it should be. We
are not told what sounds good but rather
what is wrong with the sound of just
about everything out there except maybe
one of those $150,000 all-out high-end
systems. There is far too much emphasis

on the cost of components and how cost
is related to "sound quality," even though
most high-enders will deny it.

At one time I was going to be an au-

dio engineer. I chose instead to become a

psychologist but have never left my
scientific orientation. As a result, I have

become what might be termed a psychol­ogy critic. I have remained true to only
empirically based studies of behavior and
have taken many courses in research de­sign and statistics. There are some inter­esting parallels between psychology and
high-end audio, the most obvious being
the lack of empirical support for the as­sertions that are so commonly made. Psy­chology always has been and continues to
be filled with interesting but often worth­less theories that become the basis for in­teresting but often worthless therapies.

As in high-end audio, the public can spend

hundreds of thousands of dollars on tech-

nologies (therapies) that are of dubious

value.

I had become so frustrated with the
audio scene over the last year or so that I
was hardly even reading anything any­more. In trying to purchase some audio
products during this same time, I was
convinced to buy some products by an
audio salesman, which turned out to be a
big mistake. Luckily the store that had

sold them to me was happy to give me
my money back when I returned them un-

happy. However, this was for accessories

like cables, not big-money items like
speakers, amplifiers, or preamps. I even

got into a big argument with a salesman

about the purchase of a Toslink cable for

copying CDs onto DAT. I was told that

several other products were far superior
and "what I really wanted was..." What I

really wanted was a Toslink cable! Dur-

THE AUDIO CRITIC

pdf 7

ing the argument, I heard things like

"...but according to Robert Harley...." I

am not an engineer but I was skeptical of
Mr. Harley's qualifications based on
many statements he had made in Stereo-
phile. The salesman exhibited his own ig­norance when it became clear that he did
not know what a gound loop was. I be­came so angry at the salesman's persis­tence and ignorance that I left the store
and returned later to buy my Toslink
cable from another salesman (this was the
only store in town that carried the cable).

Some of your readers have written
to suggest that you cease your "tweak
bashing" and "ignoramus hunting." I
have written to both The Absolute Sound
and Stereophile over the years to criticize
their views on various topics, but mostly
to point out their lack of understanding of
scientific methods (mostly Stereophile).
The Absolute Sound published two of my
letters but "disqualified" my point of
view in print by pointing out how inferior
my equipment was and claiming that I
would not be able to hear the things they
were talking about with such equipment.
Stereophile has never published or ac­knowledged any of my letters. Like The

Audio Critic, I have tried to write letters

to Stereophile that were "unanswerable,"
after my experience with The Absolute

Sound.

The Audio Critic appears to be the

only publication I am aware of that takes

a truly serious approach toward the eval-

uation of so-called high-end components.
Given the prices some of these compo­nents currently have, such a serious

approach is really needed. Just as impor­tantly, given the pervasive misconcep­tions about high-end audio components

that exist today, the "tweak bashing" and
"ignoramus hunting" in The Audio Critic
probably does not go far enough. I'm not
suggesting that more attacks are needed,
but that more clarification is needed and
somehow outside of this one publication.
Audio should not be the "sport" that An­thony Cordesman so often refers to it as
(sport in that context seems to denote the
constant trading of usually expensive
equipment in the never-ending quest for
perfection, and the fun of debating the
theoretical issues) but rather a means to
enjoy music outside the concert hall with
as much realism as possible at an afforda­ble price and without having to have a de­gree in electrical engineering. As an

aside, I can no longer wade through what

I call the "high-end babble" that so per-

ISSUE NO.21 • SPRING 1994

vades Mr. Cordesman's reviews in Audio.

Sincerely,
Chuck Butler
Kalamazoo, MI

Your comparison of psychology and
high-end audio is right on the money.
You haven't told us, however, how a lay­man in search of effective therapy can
distinguish, and navigate, between the so-

phisticated empiricists and doctrinaire

theoreticians in your profession—it's a
tough question, isn't it? In audio, the an­swer to the analogous question is implicit
in your complimentary remarks: every

man, woman, and child who owns more

than three CDs should have a subscrip­tion to The Audio Critic, right?

—Ed..

The Audio Critic:

Tom Nousaine, for whatever reason,
seems incapable of accurate reportage
about me or my views. Having given to

the readers of The Audio Critic a distort­ed picture of our not unpleasant encoun­ter at the Stereophile High-End Show last
year, he then compounds his errors by
perverting some perfectly clear state­ments from Professors Greiner and Lip­shitz on the audibility of absolute polari­ty. I beg to set the record straight on these
latter facts.

"[Lipshitz's] results were significant
only when trials using test tones were in­cluded in the analysis," Nousaine says.
On the contrary! Listen up:

[Here follows a series of referenced
quotations from the writings of Lipshitz,
Greiner, Richard Heyser, et al., with
pithy anti-Nousaine comments by the let­ter writer. The trouble is that the quota­tions are heavy-handedly selected, ex­cerpted, and edited with massive ellipses,
omitting the qualifying words, phrases
and sentences, falsifying the context as
well as the chronology, and not contra­dicting Nousaine's highly specific state­ments at all. It would take an additional
page, or more, to include this obviously
manipulative "documentation" here, and
I refuse to do so.—Ed..]

...Such results [as quoted above]
are difficult for those in the Nousaine
camp to swallow, for commonly they
judge others by what their own ears can
hear, or cannot. Always a mistake. Yet
one may still enjoy beholding their ver­bal contortions as they chew the truth
about polarity served up by every legiti­mate researcher on record. Dead to rights,

we have them here: in denial, every one,
about a fabulous free fix. Will Tom throw
in the towel at last? How about all the au­dio critics he has helped lead astray? Stay
tuned! Find out whether "the muffling
distortion" (my term) will later be pro­claimed over this same station.

Finally, regarding reader Donald
Scott's recent letter dissing the under­signed's "cranky" advocacy of polarity as
"the cow chip effect": the errors in his
experiment to disprove polarity are so
rife and obvious, I must invite the poor
soul to write for a free copy of my ex­planatory book The Wood Effect, and
that's no bull.

Clark Johnsen
The Listening Studio
Boston, MA

/ have made an exception here to my

"no further soapbox opportunities for

Clark Johnsen" policy as stated in Issue

No. 18 because your letter—in which
you're up to your usual tricks of creative
documentation—provides me with an op-

portune lead-in to unedited quotations

from Lipshitz and Greiner for the benefit

of our readers, showing them what these

researchers really meant.

The following is an uninterrupted
and unedited quotation from S. P. Lipshitz,
M. Pocock, and J. Vanderkooy, "On the
Audibility of Midrange Phase Distortion
in Audio Systems," J. Audio Eng. Soc. 30
(September 1982): 580-95 (your own fa­vorite reference).

...On normal musical material
heard via loudspeakers in an aver­age listening room, we have not
thus far detected the effect of mid­range phase distortions of up to
two cascaded all-pass networks of
Q 2 2. We do not have evidence
to conclusively demonstrate wheth­er phase distortions of this amount
can be heard in normal reverberant
loudspeaker listening to normal
musical or acoustic transients
(which are largely oscillatory in
nature), but it is clear that the ef­fect, if audible, is extremely subtle.

More work needs to be done in
this area, so that transducer design­ers can make an intelligent deci­sion on the significance (not the
existence) of phase effects.

Finally, we wish to caution
most strongly against quoting our
results out of context. All the ef­fects described can reasonably be
classified as subtle. We are not, in
our present state of knowledge, ad­vocating that phase linear trans­ducers are a requirement for high­quality sound reproduction. More

7

pdf 8

research is necessary. We do not
wish the research outlined above
to suffer the same misunderstand­ings, distortions, and misapplica­tions as have occurred in recent
years with transient intermodula­tion distortion. We feel that listen­ing rooms will become more
anechoic as more sophisticated re­production systems become avail­able. Thus the increased audibility
of the phase effects which we have
found with headphones may in the
future apply also to loudspeaker
listening.

So much for what Lipshitz, unfiltered

through Johnsen, wants us to understand.
As for Greiner, here is an uninterrupted
and unedited quotation from R. A. Grein­er and D. E. Melton, "A Quest for the Au­dibility of Polarity," Audio 77 (December

1993): 40-47.

While polarity inversion is not
easily heard with normal, complex
musical program material, as our
large-scale listening tests showed,
it is audible in many select and
simplified musical settings. Thus,
it would seem sensible to keep
track of polarity and to play the
signal back with the correct polari­ty to insure the most accurate pos­sible reproduction of the original
acoustic waveform.

Authors' Addendum: The work
presented here was done in 1991.
(It is now September 1993.) Since
then, there has been some, but not
much, progress made in establish-

ing polarity standards in the re­cording industry. This work is con­tinuing at the present time. There
has been some discussion in hi-fi
publications and much anecdotal
reporting, in various publications,
on the audibility of acoustical po­larity inversion. There has been
nothing noteworthy in the profes­sional literature, however, that cla­rifies the issue or "proves" that au­dibility of polarity inversion is a
major factor in listening enjoy­ment. While it is not clear why this
is the case, several factors might
be: The difficulty of doing the ex­periments in a controlled way, as
evidenced by this work; the fact
that the effect of polarity inversion
is small in most program material,
or the fact that the effect seems to
be small compared to the many
other variables in the recording/
reproduction processes (micro­phone use, room acoustics, elec­tronic processing, and the like).
Nevertheless, it seems reasonable
that at some point another step to­ward achieving greater audio
fidelity will be maintaining polari­ty of the signals throughout the
record/reproduction chain.

To sum up, what Lipshitz and Grein-

8

er are really telling us is: yes, it's better
to pay attention to polarity than to ignore
it, but no, it isn't a big deal from a listen­ing point of view. In other words, Clark,

the horse you are trying to ride to the
higher reaches of the audio world, while
a real horse and not a donkey, is a rather
slow and somewhat lame mount, unlikely
to get you there. Why don't you trade it in ?

Lastly, the perpetrator of the phony

"triple-blind" listening test (see Issue
No. 17, pp. 44 and 47) is in no position to
reprimand Donald Scott or anyone else
about experimental errors. Get your own
act together, Clark, and until you do,

please don't write us again. (Maybe you

should move to Warsaw, where you could
really experience Absolute Polarity.)

—Ed.

The Audio Critic:

.. .I have been pleased by the profes­sionalism of your reviews, and I am re­newing [my subscription] primarily be­cause I am interested in seeing how your
publication evolves with the contribu­tions of your new editors. I have more
than enough things to read in my profes­sional life, and I am less interested in rig­id publication schedules and first-class
mailings than I am in high-quality, ana­lytic criticism. If you were to conduct a
reader survey, I would give the follow­ing answers:

I became aware of your publication

from an ad in Audio. I subscribed because

I had received a gift subscription to Ste-

reophile and I felt that I needed an anti-

dote to the logorrhea and what you refer
to as "tweakism." That subscription has
now lapsed, and I am continuing with
you, partly to understand better which
parts, circuits, and features are essential
to quality construction and which are not.
Most of my equiptment was purchased
between '77 and '82 and is still perform­ing reliably. I am not "looking for some­thing to buy."

I have followed with some interest
the pleas for "tolerance of opposing
views" and finding "a halfway ground" of
agreement. I am a Radiation Oncologist. I
would not recommend any course of ther­apy without first evaluating prospective
double-blind studies. This is how I gener­ate my professional opinions, and I will
not give any weight to the opinions of
those who do not go through a similar
process. Subjectivism in medicine can be
deadly. It has no place in any of the phys­ical sciences. Some understand this; some

do not. I do not tolerate unprofessional
opinions in my field and would never ask

you to do so in yours.

Vincent Capostagno, M.D.
Merced, CA

Readers of The Audio Critic are

rather sharply divided into the categories
of those who want to learn something
(such as you) and those who want to buy
something. We try to satisfy both mind­sets; what we refuse to do (although
there is a demand for it) is to rattle off a
long laundry list of buy-this, don't-buy­that recommendations without document­ing the reasons.

I would like to expand on your

"some understand this, some do not" ob­servation regarding scientific objectivity.
I have come to the tentative conclusion
that some have a natural gift for unde­standing the basic essence of the scien­tific method, requiring perhaps only a
good junior-high-school course in Gener­al Science to assimilate the idea, and that
others are color-blind or tone-deaf, so to
speak, to the scientific mode of thinking,
although otherwise intelligent and re-

peatedly exposed to the best scientific

influences. That would explain the im­mense stubbomess and impenetrability of
some far-from-stupid members of the
audio community on the subject of "I can
hear it" vs. "I can prove I can hear it,"
when the distinction is obvious to enlight­ened twelve-year-olds.

—Ed.

* * *

/ also want to respond here collec-

tively to a whole bunch of correspon-

dence—long, well-written, laser-printed

letters as well as illegibly scribbled,

barely coherent ones, both long and
short—in which the common theme is
audible differences between various piec­es of equipment and the common failure
is a disregard for the need to eliminate
observer bias and the placebo effect.

There is no point in publishing any of

these letters; we would just be going
around in circles, repeating over and
over again that without blind listening
tests at precisely matched levels all such
discussions are meaningless. How many
times do we have to reiterate that self­evident precept before it sinks in, without
any of these "yes, but" arguments? Or is

there anyone out there who actually be­lieves that unmatched levels and peeking

at the nameplates will get you closer to

the truth? That I can't deal with. •

THE AUDIO CRITIC

pdf 9

ISSUE NO. 21 • SPRING 1994 9

In Your Ear

pdf 10

A Moderately Technical Tutorial for the Serious Audiophile

Clock Jitter,

D/A Converters, and

Sample-Rate Conversion

By Robert W. Adams

Analog Devices, Inc., Wilmington, MA

Forget everything you have read in the "alternative" audio journals
on the subject of digital jitter and start from scratch here with the
correct scientific foundation.

Foreword by the Tech. Ed.:

"The Jitter Game"

I have used the same title for this
foreword to an important article on
jitter as Stereophile's Robert Harley

uses for his articles on jitter, but with
a different meaning. Harley is play­ing a game of pretend engineering
when he attempts to analyze the jitter
of a CD player and correlate the re­sultant measurement to the sound
quality he perceives. Why does Har­ley spend so much time on jitter? Be­cause he thinks that it strongly corre­lates with the sound quality of the
equipment. Since open-loop (i.e.,
nonblind) listening tests are subject
to externally originating listener
bias, it is easy to see how he can de­lude himself to arrive at such a con­clusion. Stereophile is unfortunately
quite influential, and jitter has thus

become in the early '90s what TIM
(transient intermodulation distortion)
was in the late '70s and early '80s.

But Harley has a huge problem
because clock jitter cannot be mea­sured directly at the output of a black
box. The effects of jitter can be as­sessed indirectly from black-box
measurements, but in a correctly de­signed CD playback system these
effects are commingled with, and

usually swamped by, noise and dis­tortion products. Indeed, in exotic de­signs, the loony-tune analog stages
are so riddled with noise and distor-

tion that even large amounts of jitter
would have little effect on the mea­surements. To overcome this prob-

lem, Harley plays his little game. He
takes off the cover, gets inside the
unit, and attempts to measure the jit­ter on the internal clock line. Now,
two problems exist when he does
that: (I) since jitter is an internal pa­rameter, its effect on the external

performance of the system is depen-

dent on other aspects of the system's
design, so it is not possible to com-

pare the measured results directly

between two models under test; and
(2) measuring clock jitter is a non­trivial task, subject to many errors
even when conducted by one skilled
in the art.

Note that Harley could continue
to play his game and make other
measurements while he has the cover
open, such as I/V settling time, pow­er-supply rejection ratio, the amount

of closed-loop feedback, power­supply output impedance, etc. All

these parameters could affect the

sound quality, and under Harley's

rationale—that being unable to ob-

serve the effect of such parameters at

the output of the system does not

mean they do not affect the sound
quality—one could ask why he
doesn't make these additional mea­surements. My contention is that
Harley would indeed make these
measurements, and then delude him-

self into thinking they were remark­ably revealing of sound quality, if
some manufacturer delivered to him
a test system for a given parameter
and showed him step by step how to
use it.

Jitter and its effect on the perfor­mance an electrical system is a
difficult subject, truly understood by
only a few experts involved in the de­sign of systems sensitive to jitter. As
a result, much misinformation on jit-

ter has been circulated in the press,

originating from manufacturers' press

releases reproduced without any
competent review. In an attempt to
clear the air on the subject, we com­missioned an article by a genuine ex-

pert in the field of digital audio, Rob-

ert W. Adams, of Analog Devices,
Inc. This article is based on a paper
Adams presented at the 95th Conven­tion of the Audio Engineering Society
in New York last October. (The pre-

print number was 3712.) Bob Adams

is perhaps the youngest Fellow of the
AES (the highest honor awarded in
the field of audio engineering), and
his many pioneering achievements in
digital audio at AD, and before that
at dbx, are too numerous to be sum­marized here. His investigations in
the field of jitter reduction have re­sulted in a new method to attenuate

jitter, a practical asynchronous sam­ple-rate converter chip, which is ex­plained in his article. Before this

10 THE AUDIO CRITIC

pdf 11

chip, asynchronous sample-rate con-

verters could be had only at very

high prices and in many cases did
not perform very well. Since the new

Analog Devices ASRC chip is all-

digital, it offers the potential for the
easier and cheaper implementation
of a jitter attenuator than a multiple

phase-locked-loop S/PDIF decoder.

As I said, this is not an easy sub­ject, and the article below is not sim­ple. The problem again is that we

are trying to explain how an internal
system parameter affects the total
system performance. If Harley had
not made jitter his hobbyhorse, we
might not have found it necessary to

run such a complex article. But given
the current trendiness of jitter in au­dio journalism, I think it is important
that the serious audiophile try to go
through the article in order to separ­ate the facts from the fictions the
high-end charlatans are trying to le­gitimize. If nothing else, this article
will acquaint you with the complex

interrelationships involved in the dig­ital design process. Note that while
Bob Adams has simplified as much as
was possible without leaving out the
essentials, anybody who attempts to
measure the performance of an
S/PDIF decoder, let alone design
one, had better have a much more
complete knowledge of the subject.

Indeed, it is not possible to read the
professional literature in this field
without a strong background in sig-

nal processing, modulation theory,
and random process.

Unfortunately, it is not clear

whether Robert Harley understood

Adams's AES paper from which the
present article is derived. In the Jan-

uary 1994 issue of Stereophile he
wrote that "the paper stated that a
converter's jitter sensitivity is a func­tion of the clock frequency and over­sampling rate. This conclusion con-

firms the validity of our technique of

expressing clock jitter as a propor­tion of the clock frequency." Read

Adams's simplified but thorough ex­planation below and judge for your­self whether that's what he is really

saying.

This article does not discuss the
S/PDIF encoding and decoding pro­cess itself; that discussion is planned

for a future issue.

—David Rich

ISSUE NO. 21 • SPRING 1994

0 Introduction

Although clock jitter has re­ceived a great deal of recent attention
in the popular audio press, its effect
on signal fidelity is poorly under­stood by most journalists, and many
inaccurate statements have appeared
in print. The purpose of this article is
to introduce the fundamentals of
clock jitter and to demonstrate how
it actually affects final signal quality
for various types of D/A converters.

We also will cover an exciting
new development in sample-rate con­version and show how it will influence
the next generation of digital audio
equipment.

It has become popular practice to
measure clock jitter in commercial

outboard D/A converters using an
FM demodulator attached to the

clock pin. The output of the demodu-

lator is fed to a spectrum analyzer,

so discrete components present in the

jitter waveform may be analyzed.

Unfortunately, the amount of degra-

dation a particular jitter spectrum will

cause in the output signal depends on
the type of D/A converter used. To
interpret the results of such a mea-

surement, one has to take into ac-

count at least the following signif­icant variables:

(a) Converter type—conventional

resistive ladder, sigma-delta, or
MASH.

(b) Clock frequency applied to

the converter.

(c) Output filter type—switched-

capacitor, active RC, or a combina­tion.

(d) Any digital dividers between

the measured clock pin and the inter­nal D/A clock rate.

(e) Interpolation ratio.
(f) The frequency and amplitude

of the input signal.

(g) Jitter introduced internally to
the D/A converter (not measurable
except by its effect on the signal it­self).

These variables have more than
a minor effect on the jitter sensitivity.
With the worst combination, phase

jitter may have to be lower than 20

ps rms to obtain signals of 16-bit
quality, as opposed to more than 1

ns for the best case. Clearly, the rela­tionship between clock jitter and the
analog output is complex enough that
one should understand the fundamen-

tals before making any judgments
based on jitter about the quality of a
particular piece of equipment.

Another complicating factor will
soon be introduced commercially: a
new chip from Analog Devices
called an "asynchronous sample-rate
converter," rapidly making its way
into outboard D/A processors. This
chip acts as a universal digital buffer
between an input at one sample rate
and an output at any other sample
rate. As a byproduct of the algorithm
employed in the chip, jitter on either
the input or output sample clocks is
largely eliminated. While most engi­neers understand how a conventional
analog PLL may be used to remove
clock jitter, it is not obvious how an
all-digital sample-rate converter can
accomplish the same task. Later in
this article we will discuss how use
of this chip affects jitter in D/A con­verters.

1 Review of Clock Jitter

Clock jitter may be defined as

the time displacement of a clock sig-

nal relative to an ideal clock signal

with no jitter. Note that all the infor­mation about jitter is contained in the
edges of the clock signal, and it is
common to specify jitter in the time
domain as either the p-p or rms devi­ation of any edge from its ideal posi­tion over many thousands of clock
cycles. Most digital systems will
change state only on one edge of the
clock signal (the rising or falling
edge), in which case the jitter is mea­sured on the clock edge to which the
system responds.

In practical systems, the com-

mon types of clock jitter are:

(a) Random variations in the ar­rival of clock edges relative to their
ideal positions. For advanced read­ers, this type of jitter is referred to as
white phase jitter, as it may be pro­duced by feeding a random-noise
(i.e., white-noise) signal into a phase­controlled oscillator.

(b) Random variations of the
width of a clock pulse. This type of

jitter differs from (a) in that each

edge is referenced to the previous
edge rather than to a hypothetical
ideal clock signal. Again, for E.E.
types, this jitter is referred to as
white FM jitter, as it can be produced
by feeding a white-noise signal into a

11

pdf 12

frequency-controlled clock generator.

(c) Correlated variations in the
clock edge events relative to an ideal
clock. By correlated we mean that
the instantaneous time displacement
measured on each clock edge is not
an independent event but is in some
way related to previous clock edge

times. For the technical reader: this
causes a jitter "spectrum" which is
nonwhite and may have spectral
peaks at particular frequencies. We
will refer to this type of jitter as cor­related jitter. If the variation in clock
frequency is "slow" compared with
audio frequencies, we will call this
low-frequency correlated jitter; if
these variations are fast compared
with the audio spectrum, we will call

this high-frequency correlated jitter.

2 The Pitfalls of Time-Domain
Measurements

It is common to estimate clock

jitter by using an oscilloscope with a

very accurate time base. This prac­tice is dangerous, as the results ob­tained depend on the type of jitter as
well as on the measurement tech­nique. It also is often the case that the
oscilloscope used will have more jit­ter in its time base than is present in
the clock itself. Advanced instru­ments are available to make accurate
measurements of jitter but are not
used enough.

Figure 1 shows one measurement
technique where an oscilloscope is set
to trigger on a clock edge and the time
base is set so that only the next edge
is visible on the scope. The variations
in the arrival time of the later edge

can be used as a measure of p-p jitter.
More sophisticated oscilloscopes can
plot a histogram of zero-crossings,
allowing a more accurate estimate of
the rms jitter without resorting to

"eyeball" measurements.

Since we are triggering on one

edge and measuring the arrival time
of the next, we are assuming that the
first edge (the one we are triggering
on) is in its "ideal" time position.
This technique is fine if we are
measuring white phase jitter as
defined above, where the errors in the
clock edge positions do not accumu-

late over time relative to an ideal

clock signal. But suppose that the fre­quency of the clock signal is slowly

wandering by a small amount (low-

12

frequency correlated jitter). This

slow wandering of the frequency
causes large displacements of the
clock edges relative to a stable clock,

but the edge-to-edge measurement
technique will not reveal this effect,
as each measurement is made in ref­erence to the last clock edge only.

Another common technique is to
trigger an oscilloscope on a clock
edge and, by using the delayed trig­ger feature, examine the edges that
occur at some later time (for exam­ple, 10 clock edges later). See Figure

2. If we assume that, again, we have
a clock signal with slowly varying
frequency, we can see that this mea­surement technique will start to re­veal this low-frequency jitter compo­nent as long as the trigger delay is
long enough for the frequency of the

clock signal to have changed substan­tially between the moment when the

trigger event occurred and when the

delayed edge is examined some time

later. But one danger of this tech-

nique is that it is possible the jitter
frequency is correlated in such a way
that at particular trigger-delay values
the delayed edge of the clock has re­turned to its correct position. This
technique therefore has periodic oc­currences of "blind spots" relative to
the modulating frequency of the
clock generator and is not to be trust­ed if the clock signal contains highly
correlated jitter components.

The predominant type of noise
mechanism present may be estimated
by examining a succession of de­layed edges and observing how the

jitter behaves as a function of delay

time. White FM jitter as defined
above will display a square-root rela­tionship between delay time and ob­served edge jitter, as each clock peri­od is an independent jitter event, and
therefore many such events add in
rms fashion. Jitter which contains
low-frequency modulation of its fre­quency will show a linear relationship
between delay time and observed jit­ter. White phase jitter shows no in­crease in observed jitter with trigger
delay time, as the errors do not accu­mulate over time. Correlated jitter
shows a more complicated relation­ship between edge-to-delayed-edge
delay times and observed jitter.

The discussion above indicates
that time-domain jitter measurements

are dangerous, although useful infor-

mation may still be obtained if one is
careful. It is preferable nonetheless to
use a high-quality FM or PM detec­tor in conjunction with a spectrum
analyzer, provided one knows how to
interpret the results [Robbins 1982].

3 Sources of Jitter in Practical

Clock Circuits

Consider the clock circuit of Fig­ure 3, which is a typical RC oscilla­tor, such as might be found in the
voltage-controlled oscillator used in
a PLL. The VCO works in the fol­lowing fashion. Assume at the outset
that the capacitor is charged to V

ref

Low and the logic has turned on the

switch which connects the current
source Iup to the capacitor. The volt-

age on the capacitor now rises with a
slope proportional to Iup. When the
voltage on the capacitor reaches V

ref

Hi, the upper comparator changes
state and the current I

down

is now
connected to the capacitor, causing
the capacitor to begin charging down
at a rate proportional to the current.
When the voltage reaches V

ref

Low,
the lower comparator changes state,
and the whole operation repeats it­self, forming an oscillator.

From the previous explanation,
we can see that the current I deter­mines the frequency of oscillation of
the oscillator.

There are at least three possible
sources of jitter in this circuit. Here

we are analyzing only the one

caused by thermal noise. Practical ef­fects such as correlated frequency

components on the power supply or
those picked up because of magnetic
coupling will of course be added.

The first source of error is noise

in the current that charges and dis­charges the timing capacitor C1.
Since this current directly controls
the frequency of the oscillator, noise
on this current source translates di­rectly into white FM clock jitter.

The second source of error is

thermal noise at each comparator in­put, which causes the comparator to
switch at the wrong time. Since each
pulse is referenced to the end of the
last pulse, any variation in clock ar­rival time will be "remembered" by
all subsequent pulses, and therefore
this mechanism must again produce
white FM jitter. This can be verified

THE AUDIO CRITIC

pdf 13

Jitter Sources:

1) Current source noise (makes FM jitter).

2) Comparator noise (makes FM jitter).

3) Buffer intercept noise (makes phase jitter).

Output

Figure 3: Jitter sources in a typical RC oscillator.

Vref HI

Vref LOW

Figure 2: Edge-to-delayed-edge jitter measurement.

Scope Display

Delayed sweep

p-p jitter

Scope triggers hers.

Scope Display

Jitter

Scope triggers here.

Ideal non-jittered
output

Jittered Output

Figure 1: Edge-to-edge jitter measurement.

ISSUE NO. 21 • SPRING 1994 13

pdf 14

by imagining that the comparator
noise is dc (equivalent to an offset).

It is easily seen that this causes a

shift in frequency.

The third source of error results
from thermal noise on logic gates
that are fed with finite rise-time clock
signals. Such noise on the inputs of
these gates is translated into jitter
from the delta-t to delta-v conversion
that occurs due to the finite rise time
of the input signal. Since this mecha­nism has no "memory," it results in
white phase jitter.

There are several other types of
oscillators that offer improved jitter
performance. Crystal-controlled os­cillators are best. Voltage-controlled

crystal oscillators are available, and
these are sometimes used to recover
the clock from the incoming serial
data stream (from the output of a CD
player, for example). While they
have low jitter, they suffer from a
limited frequency-adjustment range
(about 0.1% maximum). Varactor­tuned LC oscillators are better than
the RC oscillator described earlier,
but not nearly as good as a crystal os­cillator.

4 Sensitivity of D/A Converters
to Clock Jitter

The effect of clock jitter on vari-

ous types of converters is complex

and depends on many factors. Con­verter topologies may vary in their

sensitivity to jitter by several orders
of magnitude, depending on the na­ture of the jitter.

For the purposes of analyzing jit­ter sensitivity, D/A converter fall into
three classes.

(a) Conventional, resistive lad­der converters with or without inter­polation filters.

(b) Sigma-delta converters with
continuous-time output filters.

(c) Sigma-delta converters with
switched-capacitor output filters.

4(a) Conventional, resistive ladder

D/A converters:

The effect of jitter on the output
of a D/A converter can be analyzed
by subtracting the output of a D/A
converter that uses a jittered clock
from the output a theoretically per­fect converter that uses a nonjittered
clock, and then looking at this differ­ence in the time domain. Figure 4

14

shows this analysis technique for the
case of a conventional D/A converter
with two different input frequencies
and no interpolation filter.

Figure 4a shows a 1 kHz sine
wave sampled at 50 kHz. The differ­ence between jittered and nonjittered
D/A outputs is seen to be a series of
narrow spikes whose width is propor­tional to the instantaneous difference
between the arrival time of the ideal
clock and that of the actual jittered

clock, and whose height is propor­tional to the change in signal ampli­tude from the previous to the current

sample. Here we show the case for
white phase jitter, which does not ac-

cumulate over time. Note the modu­lation of the error spikes by the sig­nal slope, which causes the error to
become very small at the signal
peaks.

We are now in a position to ana­lyze the added noise and how it re­lates to the signal. The spectrum will
be white, as there is no statistical re­lationship between one error pulse
and the next. The rms amplitude of
the noise spectrum is related to the
average slope of the DAC output sig­nal, as large step sizes between adja­cent samples cause large error pulses
in the error signal. This fact can be
seen clearly in Figure 4b, where a
higher-frequency signal (6 kHz) has
been applied to the D/A converter,

resulting in larger step sizes between

adjacent samples and hence larger er-

ror pulses.

We can summarize by saying
that for white phase modulation of
the clock, the D/A output will be
corrupted by white noise whose rms
amplitude varies with the average
slope of the signal. The worst-case
signal for audio would therefore be a
full-scale 20 kHz sine wave at the
D/A output.

The situation is slightly different
when an interpolation filter is used in
front of the D/A converter. Analysis
of that is beyond the scope of this ar­ticle, but the results are simple: the
sensitivity to white phase jitter is re­duced in direct proportion to the
oversampling ratio. This means that a
D/A converter using a 16x interpola­tion chip will be four times less sen­sitive to jitter than one using a 4x
oversampling filter (assuming that
the absolute jitter in ps is the same

for both clocks).

If the jitter is not white phase jit­ter but rather a relatively slow varia­tion of the clock frequency (low­frequency correlated jitter), then the
situation is quite different. Assume
that we feed the DAC with a sine
wave. Spectrally speaking, a slowly
wandering clock signal will cause
narrowband noise "skirts" to appear
around the frequency of the sine
wave signal. Oversampling no longer

has much effect on the output spec­trum, as the errors introduced by the
clock modulation are all "inband"

(below 20 kHz).

Many types of jitter fall in be-

tween the pure white phase jitter and
slow frequency-variation type of jit­ter described above. In that case,
oversampling may improve the jitter
sensitivity to a certain degree, but not
as much as in the case of truly ran­dom white phase jitter. Jitter in
which the time base is sinusoidally
modulated will potentially produce
discrete frequency components spaced
around spectral sticks in the DAC

output signal.

For resistive ladder converters, it
is obvious that with no input signal
(or dc), jitter cannot have any effect
on the output. The output is not
changing, so it doesn't matter exactly
when it doesn't change! While this
observation may seem trivial, the
same statement cannot be made for
other types of converters, as we shall
see presently.

In summary, regarding resistive
ladder D/A converters, we can state
the following:

• For white phase clock jitter, the

jitter spectrum on the DAC output is

white and proportional to the aver­age of the absolute value of the sig­nal slope. Oversampling filters de­crease the jitter sensitivity in direct
proportion to the oversampling ratio.
With no input to the DAC, jitter has
no effect and does not raise the noise
floor.

• For "slow" variations in the
frequency of the clock signal, narrow
noise sidebands appear around sinu­soidal components in the D/A output
spectrum, again with an amplitude
proportional to the frequency and
amplitude of the sinusoid. Oversam­pling filters do not decrease the jitter
sensitivity in this case.

THE AUDIO CRITIC

pdf 15

Figure 4

Figure 4a:
Time-domain jitter error
for a 1 kHz signal.

Figure 4b:

Time-domain jitter error

for a 6 kHz signal.

ISSUE NO. 21 • SPRING 1994 15

pdf 16

Figure 5: Time-domain jitter waveforms for 1-bit D/A converters.

4(b) One-Bit Noise-Shaping D/A
Converters with No Switched-Cap
Output Filter

This type of converter has be­come very popular in recent years,
both because of its inherent linearity
and because it can be implemented in
an all-digital CMOS process.

One-bit noise-shaping converters
can be further divided into two class­es. The first is a single-loop modula­tor with 1-bit quantization, and this
1-bit signal is fed directly to a 1-bit
output stage at the modulator clock
rate. The second class of converters,
so-called MASH converters, involve
multiple loops with feedforward
noise cancellation. They internally

produce a multiple-level digital sig­nal, which is converted to a 1-bit sig­nal through digital pulse-width mod­ulation. To achieve the desired time
resolution for the digital pulse-width
modulator, a clock signal with a very
high frequency is typically used.

1

In the previous section, we saw
that the sensitivity to clock jitter was
proportional to the changes in the
DAC output from one sample to the
next. For 1-bit converters, this
"change" is always full-scale, regard­less of the actual input signal!

Figure 5 shows a 1-bit waveform

with and without clock jitter, and the

resulting error pulse, for the case of
uncorrelated jitter (white phase­modulation jitter). Differing from the
previous case, the effect of jitter is
largely signal-independent, and in
fact the input signal to the noise­shaper loop may be zero with no re­duction in jitter sensitivity. This is
because the modulator is still switch­ing vigorously even when the input
is zero. In some cases there may be a
slight reverse sensitivity to signal
amplitude, as the number of transi­tions per unit time in the output bit­stream generally decreases as the sig­nal approaches the maximum level
in either positive or negative direc­tions.

A rough estimate of the jitter

1

The new DAC devices of NPC use
single-loop modulators with multilevel
quantizers. As in the MASH devices, the
multilevel digital signal is converted to a
1-bit signal through digital pulse-width
modulation. The new Philips-designed
$400 Marantz CD-63 uses an NPC DAC.

—David Rich

16

sensitivity can be obtained simply by
taking the average of the absolute
value of the instantaneous jitter (in
ps, for example) and dividing by the
period of the high-frequency clock
that drives the 1-bit output stage, and
then dividing by the square root of

the oversampling ratio. This noise
power is then compared with the
maximum rms signal that can be pro­duced by the 1-bit output. A quick
calculation of a typical system indi­cates a jitter sensitivity on the order
of 20 ps rms for 16-bit performance!
This is more than an order of magni­tude more sensitive than for the case
of resistive ladder converters. It is
far from trivial to design an oscillator
this good; only crystal oscillators
have a chance of meeting the spec.
Elaborate test equipment is required
even to measure jitter as low as this.

Comparing two different sigma­delta D/A converters having different
clock rates by measuring the clock

jitter for each converter is not trivial.

For example, suppose that the output
stage of converter A runs at 12 MHz
while the output stage of converter B

runs at 6 MHz (half the rate of A).
Also assume that both clocks are di­vided down from a common 24 MHz
master clock signal with a certain
amount of white phase jitter. Note
that the absolute jitter in ps is the
same for both the 6 MHz clock and
the 12 MHz clock, as a digital divid­er will maintain the absolute edge po­sitions of the master clock on its di­vided outputs. The noise produced
by each converter as a result of jitter
is proportional not only to the
amount of white phase jitter but also
to how often it occurs. As a result,
the converter running at the 12 MHz
rate will produce 3 dB more total
noise than the one running at 6 MHz
(recall that each edge displacement is
an independent statistical event, and
therefore the rms noise increases by
3 dB for each doubling of the number
of error events). However, the noise
produced by converter B is spread
over twice the bandwidth as that pro­duced by converter A, and therefore
the two converters produce the same
amount of inband noise.

This analysis applies only to 1-

THE AUDIO CRITIC

pdf 17

Figure 6: Sigma-delta converter with switched-capacitor filter.

bit output DACs. If we apply the
same experimental setup as before to
a conventional interpolated R-2R
DAC, we do not get the same results.
This is because the DAC that oper­ates at the higher-frequency clock
rate is more highly interpolated,
causing the sample-to-sample differ­ences between one output sample and
the next to decrease in direct propor­tion to the oversampling ratio. As a
result, the DAC operating at the high­er clock rate would show 6 dB less
inband noise than the same DAC run­ning at half the clock frequency. This
result is different from the 1-bit re­sult because for a sigma-delta output
stage, the sample-to-sample transi­tions are always full-scale, regardless
of the clock rate or interpolation ratio.

It is clear from this analysis that

1-bit converters are very sensitive to
white phase modulation of the clock.
This can be explained in the frequen­cy domain by the observation that
phase modulation causes the out-of­band shaped noise to "fold" down
into the baseband. In the time do­main, it is intuitively clear that the
edge-to-edge jitter is what affects this
type of converter the most. Low­frequency correlated jitter of the
clock is not nearly as damaging to
the noise performance of the DAC,
although correlated sidebands around
the signal may become a problem.

One interesting aspect of this
high jitter sensitivity is that it is
difficult for manufacturers using this
type of D/A converter to meet the
high dynamic-range specs common
with today's equipment. For exam­ple, in a system that uses a conven­tional, resistive ladder D/A convert­er, the noise is measured with no
digital codes toggling, and therefore
is determined only by analog thermal
circuit noise, which may be much

ISSUE NO. 21 • SPRING 1994

lower than theoretical 16-bit quanti­zation noise (this figure of course is
not meaningful, as the noise will in­crease as soon as the music begins).
Clock jitter has absolutely no effect
under these conditions. But in a sys­tem with a 1-bit sigma-delta or
MASH converter, the 1-bit signal is
very "busy" even when the digital in­put signal is silent. Clock jitter will
most likely be the dominant noise
source, and therefore it will be
difficult for the manufacturer to
claim 110 dB of dynamic range. In
some cases an extra circuit is used to
detect "digital silence" and the 1-bit
D/A output stream is actually turned
off under these conditions to make
the numbers look good!

In conclusion, 1-bit D/A con­verters with no discrete-time output
filters are extraordinarily sensitive to
edge-to-edge jitter caused by white

phase modulation of the clock. As we
saw before, such phase modulation
often is caused by passing a clock
with finite rise time through a buffer.
One must be very careful to reduce
this edge-to-edge type of jitter to the
lowest levels possible.

4(c) One-Bit Noise-Shaping D/A
Converters with Switched-Cap Out-

put Filtering

Figure 6 shows a block diagram

of a D/A converter with switched-cap
output filter. The origin of the name
"switched-capacitor filter" is obvious
from the diagram. The switching of
the capacitor removes its memory
characteristic. A first-order analysis
of a switched capacitor shows that it
is equivalent to a resistor of value

1/fC, where f is the frequency at

which the switches change state. In
an integrated circuit, the exact value
of a resistor or capacitor can vary by
30% or more. This makes the design

of precision RC filters impossible.
The transfer function of a switched­cap filter is dependent on the ratio of
capacitors, which can be controlled
to 0.1% in an integrated circuit. In
Figure 6 we see two switched­capacitor circuits that replace resis­tors in a continuous-time circuit. If
we replace the switched-cap circuits
with resistors, we can see that the
filter in Figure 6 is a first-order low­pass.

A more exact analysis of a
switched-capacitor circuit is required
to take into account the sampling ef­fect of the switch. This more exact
analysis shows that a switched-cap
filter must be analyzed as a discrete­time system in much the same man­ner as a digital filter. It is often a

point of great confusion how a
switched-cap output filter is different
from a continuous-time active RC
filter when it comes to jitter. The an­swer is as follows. A switched-cap
filter will settle to a particular value
regardless of when the clock edge oc­curs (assuming the op-amp is fast
enough to settle completely within
one switching interval). It is a true
discrete-time system in that the dy­namic settling behavior of all circuit
voltages is not important as long as
the settled value is correct. This is
not true for the case of a 1-bit modu­lator feeding a continuous-time filter,
where an error in the switching time
of the 1-bit signal does have an effect
on the voltage at the end of the clock
period, and in fact will change the
output voltage for many hundreds of
cycles thereafter.

From this discussion we can
conclude that a 1-bit modulator feed­ing a switched-cap filter behaves the
same as a resistive-ladder converter
running at the same oversampling ra­tio, as both converters produce volt­ages that settle to the same value re­gardless of the timing of the clock
edge. This is true as long as the
switched-cap filter completely re­moves the out-of-band noise from the

1-bit modulator. If that's not the
case, then the jitter sensitivity de­pends on the sample-to-sample dif­ferential change. If the change is
dominated by the signal slope itself,
then the jitter sensitivity is un­changed from the case where the
out-of-band noise is completely re-

19

pdf 18

Figure 7: Time-domain view of sample-rate conversion.

20 THE AUDIO CRITIC

moved. If the sample-to-sample

change is dominated by unfiltered
out-of-band noise, then the jitter sen­sitivity will be in proportion to the
rms value of the sample-to-sample
changes. Note that in cases where a
switched-cap filter is followed by a
continuous-time analog filter to re­duce further the out-of-band noise,
only the switched-cap filter is useful
for reducing the sensitivity to jitter.
This fact implies that it is impossible
to predict the jitter sensitivity of a

chip-level D/A converter that has

both switched-cap and continuous­time RC filters onboard, as it is im­possible to tell how much of the
filtering is done in each section.

2

5 Asynchronous Sample-Rate
Converters (ASRC) and Jitter
Reduction

In a previous paper [Adams and
Kwan 1993] I described an algorithm
and VLSI implementation of it which
allow sample-rate conversion between

arbitrary asynchronous rates. Unlike
synchronous converters, the device
accepts external clocks at Fsin and
Fs

out

, and by performing various sig­nal-processing operations on those
clocks it is able to derive a high­accuracy estimate of the current sam-

ple-rate ratio, and this estimate is
continuously updated so as to track
real-time variations in the input or
output sample rates.

Figure 7 shows a time-domain
view of sample-rate conversion. Con­ceptually, asynchronous conversion
consists of interpolating the input
sequence to an extremely high fre­quency, which causes the amplitude
differences between adjacent interpo­lated samples to become very small.

The output resampling process then
consists of picking off the nearest in­terpolated point.

The ASRC chip described uses a
polyphase filter approach, with
65,536 unique polyphase filters of
length 64, each stored in compressed
form in ROM. This approach to rate

conversion is more efficient to imple­ment than the interpolation/decima­tion model, as unneeded interpolated
outputs are not computed. While poly­phase filtering sounds complicated, it
is actually quite a simple concept.

Every FIR filter has a particular

group delay, which defines how
much delay the filter introduces to
signals appearing on its input. A typ­ical interpolation filter might exhibit
about 600 µs of group delay, for ex­ample. Most FIR filter are designed

to be linear-phase, which means that
the delay introduced by the filter is
independent of frequency.

Normal linear-phase FIR filters
have a group delay that corresponds
to an integral number of clock cycles.
But it also is possible to design an
FIR filter which is linear-phase but
has a group delay of an integer plus
a fractional number of clock cycles.
For example, a normal FIR filter of
length 100 taps might have a group

delay of 50 samples (half its length).
But it is possible to design a linear­phase FIR filter with a group delay of

50.5 samples. Now suppose that we

had a large bank of FIR filters all
connected to the same input signal,
and each of these filters had the same
frequency response but a slightly dif­ferent group delay. When we change
the sample rate of a signal, we are ef­fectively attempting to resample the
signal at a point between the existing
sampled points of the input signal.

Using the filter bank described, we

could simply pick a filter output
whose delay matched most closely

the desired resampling point for that
particular output sample. For exam­ple, if the output clock signal (which
differs in frequency from the input
clock signal) were to fall halfway be­tween two edges of the input clock
signal, we would select the filter that
has a group delay of 50.5 input sam­ple periods.

To ensure that we have enough
possible group delays to select from,
the new AD 1890 chip uses a bank of

2

The Crystal CS4303, Burr-Brown
PCM67, all MASH chips, and all NPC
chips use continuous-time filters. An ap­plication note available from Crystal
Semiconductor on the CS4303 shows
how difficult it is to create the required
low-jitter clock. The Crystal CS4328 uses
a fourth-order switched-capacitor filter
for removal of most of the out-of-band
energy. It is followed by a second-order
continuous-time reconstruction filter which

removes most of the image signals that
arise from the switch sampling of the

switched-cap filter. The Philips 1-bit

DACs are a hybrid. They use an on-chip
first-order switched-cap filter (similar to
that in Figure 6), which removes some of
the out-of-band energy. This is followed
by an off-chip continuous-time filter.

—David Rich

pdf 19