David Looser wrote:

"Eeyore" wrote
David Looser wrote:

Mind you phase jitter caused by the mechanical parts of a poor
quality or faulty CD system *can* stress the error-correction to the
point where the playback quality is compromised)

NO. Pure horse manure. The digital signal is buffered to buggery. A bit of
jitter won't bother it. This is an example of analogue-style thinking
being inappropriately attached to digital signal paths.

It's not "horse manure". No amount of buffering will cure the problem of
recovering data from a badly jittered data-stream. It's the clock recovery
that is affected. I can tell that you've never had to design clock-recovery
circuits.

Actually .... I have been involved in a project to determine weaknesses of early
examples of precisely this kind of circuit. There certainly were problems in the
early days.

If it's *badly* jittered you have a point but I've not seen jitter greater than
a couple of ns on digital audio data streams.

Graham

"David Looser" wrote in message
...
"Keith G" wrote in message
...

"David Looser" wrote in message
...

Another interesting comment, as it appears to directly contradict your
earlier one :-)


What comment is that?

quote
In the case of my son, his kit (Technics/B&W) masks any differences
between CD and LP
unquote


OK, but I would suggest that if you are going to reference previous comments
you should include them in your post; I for one will not trawl backwards and
forwards trying to work out what you are referring to....


David, all that mastertape horse**** was dealt and dispensed with in here
years ago - here's a hint:

What "horse****" is that?

If you don't *know*, don't guess....

So enlighten me.


Difficult to refuse so polite and elegant a request, so here goes:

This is an 'audio' newsgroup primarily concerned with the
production/reproduction of sound for (mostly) pleasure purposes populated by
(a few) people with differing views which I believe can be mostly divided
into two main groups - the 'accurists' (for want of a better word) who
strive for a sound which is 'as close to the original recorded signal as
possible' and the 'realists' (like me) who seek a sound which is 'as close
to the original physical sound as possible' or at least their/my *idea*
thereof ? (It all becomes highly subjective by the time the recorded sound
reaches the listeners ears in his own listening environment when, for
example, no two people would agree on the exact same *volume setting* -
never mind anything else!)

Whatever. One thing the accurists have to fall back on is the 'mastertape'
and the gauged faithfulness to this (fidelity) was frequently thrown into
the arguments as the be-all and end-all of sound reproduction. The reason
for this is that it is fairly easy to *measure* deviation from the original
signal, rules can be made from various measurements and it therefore becomes
a useful weapon. Where it becomes horse**** in my book is when reference is
continually made to mastertapes that no-one has ever heard (or ever will)
and to events that weren't witnessed personally. OK?

All the realists have to counter this with is that a sound which may not
measure too well (and is therefore 'distorted') frequently sounds better
(and more *real*) - PW, the accurist's God, said summat like the objective
of good hifi' is/would be a 'straight wire with gain'; I say fine, but I'll
bend it until it sounds better! Here's an example of how it works which is
less than 24 hours old:

Yesterday, a very nice blokey came here to hear my Dynaco monos (which are
up for sacrifice in the most vicious 'hifi pogrom' yet) and he brought his
Bryston preamp, as I don't now have one. This meant I couldn't do anything
other than start from stone cold, which I did - the 'sound' was horrible
from the off and only started to get better as the Dynies warmed up, but the
guy was very curious about the SET amps I have here as he had never heard
one! So, after a brief stint when I had swapped his Bryston out and
substituted my Denon SS amp (using the pre-outs to drive the Dynacos) which
warmed the sound up straight away (distortion) and which the blokey had
preferred straight away, I put the Bez Chinese 300B SET on (also up for
sacrifice, as it is not of my own making) and instantly the guy sat back and
said 'Ah, that's better!'

The upshot is my Bezzer is now 8 miles from here, while he checks to see how
well it drives his Altec Lansing behemoth speakers (???) - yet another
'realism beats accuracy' triumph, I believe, but I'm not sure about how well
it will work on his kit...??

:-)

In article , David Looser
wrote:
"Eeyore" wrote in message
...


NO. Pure horse manure. The digital signal is buffered to buggery. A
bit of jitter won't bother it. This is an example of analogue-style
thinking being inappropriately attached to digital signal paths.

It's not "horse manure". No amount of buffering will cure the problem of
recovering data from a badly jittered data-stream. It's the clock
recovery that is affected. I can tell that you've never had to design
clock-recovery circuits.

Rather depends on your definition of "badly". :-)

If you look at the levels of jitter people argue about in audio mags it is
of the order of 100's ps to a ns. I'd have said is quite small in the
context of SPDIF or EBU, although they are vague about how this is
measured. I'd expect this to be easily reclocked and buffered if the
engineer knows what he is doing. In practice, my experience is that even a
decade old DAC like the Meridian 263 or 563 does this with ease.

So does a fairly cheap RX like the one in the Pioneer recorders I have been
using recently. Sample-by-sample comparisons using a player with a disc
that isn't faulty shown sample-by-sample agreement of recordings. Also
agreeing with recordings made directly to computer using a CDROM drive.

Can you define what you meant by "badly"? I can see that if the jitter is
large enough then you will get problems recovering the actual stream of
values. But I can't say I have encountered this with domestic audio kit
I've used unless the disc or equipment are faulty. No doubt, though,
someone will have made kit bad enough for this to be a problem... :-)

Of course, if the 'jitter' has significant amounts at low phase modulation
frequencies then the loop or buffer will need to alter the local clock and
this will affect the output. I can also see that an RX/DAC that doesn't
reclock but simply uses the implict clock in the data will then translate
that into phase modulation of the output. But this does not mean that the
correct series of values was not RXd. Is this what you were thinking about?
That the loop or buffer will respond to close-in phase noise by adjusting
the conversion clock frequency?

Given values for 'jitter' of the order of 100's of ps I can't immediately
see that this would cause a very significant modulation of the DAC clock
rate if smoothed with a sensible buffer and loop.


Slainte,

Jim

--
Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm
Audio Misc http://www.audiomisc.co.uk/index.html
Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html

"Bob Latham" wrote in message
...
In article ,
Eeyore wrote:

However, if those first steps had not been made we wouldn't have the
truly excellent medium we have now. It just irks me that the CD standard
wasn't say 48kHz, 18 bit.

So in your eyes then 16bit 44.1KHz is not transparent otherwise why
bother.


Cheers,

Bob.

--
Bob Latham
Stourbridge, West Midlands

My preference for 48k sampling has nothing to do with audio quality, but to
do with compatibility. CD is 44.1, DVD digital TV and digital radio is 48k
or multiples thereof. Having CD at 44.1 means that for transmission on TV
or digital radio, there has to be an otherwise unnecessary sample rate
conversion between source and destination. I know that SRCs are nowadays
audibly transparent, but nevertheless, in a professional installation, it
adds an unnecesary level of complication.

It has caused the BBC for one, considerable extra complication and expense
when designing their new Broadcasting House infrastructure, as they decided
that network radio would run their studios at 44.1k, whilst News and Current
Affairs would run their studios at 48k so as to be compatible with TV. They
consquently had to provide both 44.1 and 48k routing, and for DAB and DSAT
distribution, they had to SRC the network outputs before transmission.

It would have been so much easier if CD had run at 48k.

S.


--
http://audiopages.googlepages.com

On Sun, 6 Jan 2008 18:08:08 -0000, "Serge Auckland"
wrote:

"Bob Latham" wrote in message
...
In article ,
Eeyore wrote:

However, if those first steps had not been made we wouldn't have the
truly excellent medium we have now. It just irks me that the CD standard
wasn't say 48kHz, 18 bit.

So in your eyes then 16bit 44.1KHz is not transparent otherwise why
bother.


Cheers,

Bob.

--
Bob Latham
Stourbridge, West Midlands

My preference for 48k sampling has nothing to do with audio quality, but to
do with compatibility. CD is 44.1, DVD digital TV and digital radio is 48k
or multiples thereof. Having CD at 44.1 means that for transmission on TV
or digital radio, there has to be an otherwise unnecessary sample rate
conversion between source and destination. I know that SRCs are nowadays
audibly transparent, but nevertheless, in a professional installation, it
adds an unnecesary level of complication.

It has caused the BBC for one, considerable extra complication and expense
when designing their new Broadcasting House infrastructure, as they decided
that network radio would run their studios at 44.1k, whilst News and Current
Affairs would run their studios at 48k so as to be compatible with TV. They
consquently had to provide both 44.1 and 48k routing, and for DAB and DSAT
distribution, they had to SRC the network outputs before transmission.

It would have been so much easier if CD had run at 48k.

S.

But their analogue terrestrial service needs NICAM at 32k. Yet another
sample rate conversion.

Conversion is such a trivial matter these days that I would suggest
that it need not be considered in making technical decisions.

d

--
Pearce Consulting
http://www.pearce.uk.com

Keith G" wrote in message
...


This is an 'audio' newsgroup primarily concerned with the
production/reproduction of sound for (mostly) pleasure purposes populated
by (a few) people with differing views which I believe can be mostly
divided into two main groups - the 'accurists' (for want of a better word)
who strive for a sound which is 'as close to the original recorded signal
as possible' and the 'realists' (like me) who seek a sound which is 'as
close to the original physical sound as possible' or at least their/my
*idea* thereof ? (It all becomes highly subjective by the time the
recorded sound reaches the listeners ears in his own listening environment
when, for example, no two people would agree on the exact same *volume
setting* - never mind anything else!)

In the world of cinema sound the cinema's playback system is conceptually
divided into two parts at the source selector switch: the A-chain, which is
the soundhead, pre-amps decoders etc, and the B-chain, which is the balance
of the system, volume control, equalisers, power amps, speakers plus, of
course, the acoustic environment. There is a separate A-chain for each type
of source: optical analogue, magnetic analogue, optical digital, hard-drive
etc. whilst the B-chain is the same for them all.

Films are mixed in dubbing theatres which conform to an international
standard for frequency response, reverb time etc. and cinema B-chains are
adjusted to conform as closely as possible to that standard. Meanwhile the
A-chain is adjusted to be as nearly transparent as possible, ideally the
electrical signal leaving the cinema A-chain is exactly the same as the
electrical signal leaving the sound mixer's desk. This ensures that the
sound heard by the cinema patron is as close as possible to that heard in
the dubbing theatre. It also means that, at least as far as film sound
sources are concerned, the volume and frequency response are similar
regardless of the type of soundtrack being played (there is remarkably
little audible difference when switching back and forth between the digital
and analogue soundtracks on modern 35mm film prints)

Transferring this concept to the home Hi-Fi area, the record-player plus
RIAA pre-amp or CD player, tape machine, tuner etc. is clearly the A-chain.
Whilst the volume and tone controls, power amps and speakers (+room of
course) are the B-chain. Now I grant you that there is less standardisation
of music mixing rooms than is the case with film dubbing theatres, and there
is no standardisation of home listening rooms at all, nevertheless it seems
to me that at least for the A-chain the same principle applies, the signal
leaving the RIAA amp or CD player should be as close as possible to that
entering the monitoring chain in the mixing room. If you want to
bugger-about with the sound to make it "better" you can do that with the
B-chain.

Whatever. One thing the accurists have to fall back on is the 'mastertape'
and the gauged faithfulness to this (fidelity) was frequently thrown into
the arguments as the be-all and end-all of sound reproduction. The reason
for this is that it is fairly easy to *measure* deviation from the
original signal, rules can be made from various measurements and it
therefore becomes a useful weapon.

Well no. The reason is that the mastertape represents the sound that the
sound engineer, record producer, artist etc. agreed was what they wanted the
record to sound like.

Where it becomes horse**** in my book is when reference is
continually made to mastertapes that no-one has ever heard (or ever will)

No-one? what about the recording engineer, the producer, the artist etc?

and to events that weren't witnessed personally. OK?

All the realists have to counter this with is that a sound which may not
measure too well (and is therefore 'distorted') frequently sounds better
(and more *real*)

Well yes I understand, That's why 1950s AM radios with single-ended output
pentodes running 10% distortion at 3W sound far more "real" than anything
else! :-)

- PW, the accurist's God, said summat like the objective
of good hifi' is/would be a 'straight wire with gain'; I say fine, but
I'll bend it until it sounds better! Here's an example of how it works
which is less than 24 hours old:

Long rambling story snipped.

The upshot is my Bezzer is now 8 miles from here, while he checks to see
how well it drives his Altec Lansing behemoth speakers (???) - yet another
'realism beats accuracy' triumph, I believe,

How do you work that out?, what did your story have to do with either
accuracy or realism?


David.

"Don Pearce" wrote in message
...
On Sun, 6 Jan 2008 18:08:08 -0000, "Serge Auckland"
wrote:

"Bob Latham" wrote in message
...
In article ,
Eeyore wrote:

However, if those first steps had not been made we wouldn't have the
truly excellent medium we have now. It just irks me that the CD
standard
wasn't say 48kHz, 18 bit.

So in your eyes then 16bit 44.1KHz is not transparent otherwise why
bother.


Cheers,

Bob.

--
Bob Latham
Stourbridge, West Midlands

My preference for 48k sampling has nothing to do with audio quality, but
to
do with compatibility. CD is 44.1, DVD digital TV and digital radio is 48k
or multiples thereof. Having CD at 44.1 means that for transmission on TV
or digital radio, there has to be an otherwise unnecessary sample rate
conversion between source and destination. I know that SRCs are nowadays
audibly transparent, but nevertheless, in a professional installation, it
adds an unnecesary level of complication.

It has caused the BBC for one, considerable extra complication and expense
when designing their new Broadcasting House infrastructure, as they
decided
that network radio would run their studios at 44.1k, whilst News and
Current
Affairs would run their studios at 48k so as to be compatible with TV.
They
consquently had to provide both 44.1 and 48k routing, and for DAB and DSAT
distribution, they had to SRC the network outputs before transmission.

It would have been so much easier if CD had run at 48k.

S.

But their analogue terrestrial service needs NICAM at 32k. Yet another
sample rate conversion.

Conversion is such a trivial matter these days that I would suggest
that it need not be considered in making technical decisions.

d

--
Pearce Consulting
http://www.pearce.uk.com

That's what I would have thought, especially considering the BBC desks have
SRC at each input, but they, in their infinite wisdom, decided that they
will run studios at the sample rate of their primary input source (CDs in
the case of radio, VTRs in the case of TV) and sample rate convert for each
destination.

S.


--
http://audiopages.googlepages.com

"David Looser" wrote in message
...
Keith G" wrote


Whatever. One thing the accurists have to fall back on is the
'mastertape'
and the gauged faithfulness to this (fidelity) was frequently thrown into
the arguments as the be-all and end-all of sound reproduction. The reason
for this is that it is fairly easy to *measure* deviation from the
original signal, rules can be made from various measurements and it
therefore becomes a useful weapon.

Well no. The reason is that the mastertape represents the sound that the
sound engineer, record producer, artist etc. agreed was what they wanted
the
record to sound like.


Irrelevant - whatever the mastertape is (or isn't) is not the issue; the
'accuracy vs. realism' argument is....



Where it becomes horse**** in my book is when reference is
continually made to mastertapes that no-one has ever heard (or ever will)

No-one? what about the recording engineer, the producer, the artist etc?


More irrelevancy - they weren't present in the arguments in this group,
AFAIA....



Long rambling story snipped.


Nothing like as long as your 'cinema irrelevancies' which I snipped...



The upshot is my Bezzer is now 8 miles from here, while he checks to see
how well it drives his Altec Lansing behemoth speakers (???) - yet
another
'realism beats accuracy' triumph, I believe,

How do you work that out?, what did your story have to do with either
accuracy or realism?


Because the guy leapt at a 300B SET (over his own SS pre and PP valve monos)
which, as I'm sure you are aware, are slated endlessly in this group for
'distortion' ('broken' even...) but which I maintain provide a more
*realistic* sound - kinda proves my point, doesn't it? Asitappens, he's been
on the phone since to say (verbatim) 'I'm not getting the same, sweet sound
as you get on your Lowthers!' - I'm not surprised, I didn't think he would
but I was willing to let him try!

While I'm on: I have started a ferocious campaign to reduce clutter and am
punting no end of stuff out on eBay - I have 5 auctions on now, including a
set of both Morgan Jones 'Valve Amp' books which may be of interest to
someone he

http://cgi.ebay.co.uk/ws/eBayISAPI.d...MESE:IT&ih=017


Condition *as new*....

(Sorry to say! :-)

"Keith G" wrote


Correction:


Because the guy leapt at a 300B SET (over his own SS pre and PP valve
monos)

The (Bryston) SS pre was his; the valve monos are mine...

"Jim Lesurf" wrote in message
...

Can you define what you meant by "badly"? I can see that if the jitter is
large enough then you will get problems recovering the actual stream of
values. But I can't say I have encountered this with domestic audio kit
I've used unless the disc or equipment are faulty. No doubt, though,
someone will have made kit bad enough for this to be a problem... :-)


I've no experience in this respect with domestic kit, or with CD as source,
so I will accept the consensus here that with CD this isn't a problem. My
experience is with rotating-head tape transports and with long (many km)
transmission lines running at bit rates of several hundred Mb/sec. (Yes
transmission lines can cause jitter, as well as often having a very poor
eye.)

David.