[William Sommerwerck]

Sample rate essentially *is* bit rate, in linear PCM...

No, it isn't.

Though quantization is a form of sampling, it is not generally referred to
that way. When people say "sample rate", they're talking about temporal
sampling.

[William Sommerwerck]

We already have Blu-Ray audio...

... and DVD for years with that capability.

Right and SACD, which at least proves the point that you
can fool some of the people all the time, but not all the
people all of the time. :-)

And your opinion is based on listening to how many SACDs? Two? Three?

SACDs do, generally, sound different -- and "better" than CDs. WHY is
another matter.

[William Sommerwerck]

To my ears, Blu-ray audio sounds noticeably better than
the common run of CD sound. However...

Blu-ray audio disks are audiophile recordings, and may use better mics,
fewsd\\er electronics, less processing, etc, than Compact Disk
recordings.
At the moment, I cannot say with any objectivity that "high resolution"
recordings are definitely superior. In theory, there's no reason for
them being better, as 16/44 presumably records everything that's
audible.

Simple to resample those Blu-Ray recordings to 16/44 to compare them
properly. I can't hear a difference, and never met anyone who *claims*
they
can who could prove it.
You may be the exception of course, but I wouldn't bet money on it. :-)
Tracking at 24/96 is another matter of course.

It would be useful if you actually read and understood what I wrote.

The CD layer of most SACDs is derived from the DSD recording, and thus tells
us nothing about what went on prior to the recorder. Had you read what I
actually wrote -- rather than what you wanted to read -- you'd know this.

[Don Pearce[_3_]]

On Sun, 17 Jun 2012 13:58:04 +1200, "geoff"
wrote:

Sean Conolly wrote:
"James T" wrote in message


Higher bit rates have a real benefit, but for me higher sample rates
do not. I have so many other factors that audibly compromise my
recordings, sample rate is the least of my concerns.


Sample rate essentially *is* bit rate, in linear PCM, which is what we end
up recording and playing back for the most part. Or did you mean
"bit-depths" ?

geoff


To get the bit rate in linear PCM, you multiply the sample rate by the
number of bits (bit depth), and then by the number of channels.

d

[geoff]

Don Pearce wrote:
On Sun, 17 Jun 2012 13:58:04 +1200, "geoff"
wrote:

Sean Conolly wrote:
"James T" wrote in message


Higher bit rates have a real benefit, but for me higher sample rates
do not. I have so many other factors that audibly compromise my
recordings, sample rate is the least of my concerns.


Sample rate essentially *is* bit rate, in linear PCM, which is what
we end up recording and playing back for the most part. Or did you
mean "bit-depths" ?

geoff


To get the bit rate in linear PCM, you multiply the sample rate by the
number of bits (bit depth), and then by the number of channels.

.... which is nowhere near as concise and instantly appreciable a spec as
SR/Bits/chans .

geoff

[geoff]

William Sommerwerck wrote:
"geoff" wrote in message
...
William Sommerwerck wrote:

To my ears, Blu-ray audio sounds noticeably better than the common
run of CD sound. However...

Blu-ray audio disks are audiophile recordings, and may use better
mics, fewsd\\er electronics, less processing, etc, than Compact Disk
recordings. At the moment, I cannot say with any objectivity that
"high resolution" recordings are definitely superior. In theory,
there's no reason for them being better, as 16/44 presumably records
everything that's audible.

There is some suggestion that the difference is only in different
mastering.

That's only one possible explanation. There are others.

Could you define what you mean by "mastering"?

Anything from minimal processing through different equipment (level changes
and associated dithering), through to eq and dynamic tweaks, maybe very
subtle, or not.

geoff

[Don Pearce[_3_]]

On Sun, 17 Jun 2012 22:07:33 +1200, "geoff"
wrote:

Don Pearce wrote:
On Sun, 17 Jun 2012 13:58:04 +1200, "geoff"
wrote:

Sean Conolly wrote:
"James T" wrote in message


Higher bit rates have a real benefit, but for me higher sample rates
do not. I have so many other factors that audibly compromise my
recordings, sample rate is the least of my concerns.


Sample rate essentially *is* bit rate, in linear PCM, which is what
we end up recording and playing back for the most part. Or did you
mean "bit-depths" ?

geoff


To get the bit rate in linear PCM, you multiply the sample rate by the
number of bits (bit depth), and then by the number of channels.

... which is nowhere near as concise and instantly appreciable a spec as
SR/Bits/chans .


But it is at least correct, as opposed to yours in which you divide
instead of multiply. The results are a bit different, you will
appreciate.

d

[Scott Dorsey]

James T wrote:
Of course transients will correspond to ability to reproduce
equivalent frequencies. I understand that. But I was wondering to what
extent that type of information would be likely to exceed ye olde
20-20khz spec that is regarded as the baseline for human perception.

Not at all. All of the transient timing information is contained in the
bandwidth.

People in the audiophile community keep bringing up the subject of timing
as if somehow it's independent of bandwidth and if a higher sample rate
could improve timing accuracy through some mechanism other than higher
bandwidth.

Now.... some argument has been made that the higher channel bandwidth
might be a good thing in that it could allow us to hear more accurate
timing cues. However, nobody who has looked for such a thing in actual
listening tests have ever found it. Doesn't mean it doesn't exist, but
nobody has found it.

There seems to be a lot of gray area there. IOW, did the A/D/A tests
done with the 16-44.1 loop adequately test perception of that aspect
of human hearing. Do you know of any relevant test data?

Check the AES preprint list... there were dozens of papers in the nineties.
Ignore any of the ones from anyone associated with the Kanagawa Institute.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Scott Dorsey]

In article , Trevor wrote:
"Les Cargill" wrote in message
...
24 bit depth is good for tracking because of the additional 8 bits
headroom.

There's no converter ever made that will give 8 bits extra headroom, and
never will be even with cryogenic cooling. Of course 3 or 4 bits can still
be worthwhile.

You're looking on the wrong of the scale. You can't bring the noise floor
down, so just run the thing on kilovolt rails....
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Scott Dorsey]

In article , Trevor wrote:
"Roy W. Rising" wrote in message
Through the years of emerging technology I have concluded that usually it
is possible to store more information than current reproducers can
recover.
There's more in the groove of an old 'record' than we could play/hear with
a sharpend nail hooked to a megaphone.

Since it was originally recorded to a disk master with a "sharpened nail" as
well, any extra "information" is likely to be just as innacurate.

Not at all. Listen to an acoustic photograph vs. a modern transcription of
the same disc some time. Lower distortion, not so many internal horn
resonances today. Technology got a lot better and it still keeps getting
better.

I'm not sure that this applies in the digital world, because for the first
time with digital conversion systems, the recorder and the medium itself
are no longer the bottleneck in terms of sound quality. But ask me in
fifty years, I'll have a better idea then.

I would not be surprised to discover that in fifty years, transducers are
a whole hell of a lot better than they are today and recorders are mostly
unchanged in concept and performance.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Scott Dorsey]

William Sommerwerck wrote:
Sample rate essentially *is* bit rate, in linear PCM...

No, it isn't.

Though quantization is a form of sampling, it is not generally referred to
that way. When people say "sample rate", they're talking about temporal
sampling.

BIT RATE (bits/sec) = SAMPLE RATE (samples/sec) * WORD LENGTH (bits/sample)

If there's one thing I learned in engineering school, it's to always do
the dimensional analysis even when you think you know what the units are.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Don Pearce[_3_]]

On 17 Jun 2012 08:38:14 -0400, (Scott Dorsey) wrote:

William Sommerwerck wrote:
Sample rate essentially *is* bit rate, in linear PCM...

No, it isn't.

Though quantization is a form of sampling, it is not generally referred to
that way. When people say "sample rate", they're talking about temporal
sampling.

BIT RATE (bits/sec) = SAMPLE RATE (samples/sec) * WORD LENGTH (bits/sample)

If there's one thing I learned in engineering school, it's to always do
the dimensional analysis even when you think you know what the units are.
--scott

You forgot the number of channels, Scott. Dimensionless unfortunately,
so dimensional analysis doesn't warn you.

d

[Mike Rivers]

On 6/17/2012 4:31 AM, James T wrote:

Mike, Do you happen to have any references to those tests? That's what
I'm trying to find.

Nope, just me. Either trust me or not, or go Googling.

?? Primarily a theoretical question. I've been curious about the
prevalence of 24/96 and associated controversy. I haven't found much
info that addresses that conclusively.

There are no conclusions. It's too much of an individual
thing. Is this so hard to accept?

Several years ago my friendly local dealer set up a
demonstration.

I, and most of the listeners,
had a preference for the straight-through path, some
preferred the 44.1 kHz path, some preferred the 96 kHz path,
but nobody would have always picked one sample rate over the
other every time. These were pro engineers, which doesn't
necessarily make them any more qualified than anyone else,

And they'd have an idea what to listen for. Useful info, Mike.

It isn't really a matter of what to listen for, because not
all of the differences can be verbalized, and our hearing
isn't that consistent. I can say that I think that the low
end sounds "fuller" through one path than another, but yet I
can measure frequency response of both paths and find them
to be identical to some pretty small limits. What I can't
measure accurately, though, is what actually gets to my ear,
and what my brain does with it.

If you remember where you saw that, [positional cues] please post!

My memory is much worse than my hearing. It may have been a
discussion on a forum rather than a scientific study. If you
haven't got my drift by now, let me put it to you plainly:
I JUST DON'T CARE what technology was used to make a
recording unless it sounds bad. Then I might be curious.

I don't know that anyone has ever returned a recording for a
refund on the basis that it was made at 44.1 kHz instead of
96 kHz.




--
"Today's production equipment is IT based and cannot be
operated without a passing knowledge of computing, although
it seems that it can be operated without a passing knowledge
of audio." - John Watkinson

http://mikeriversaudio.wordpress.com - useful and
interesting audio stuff

[Randy Yates[_2_]]

"Trevor" writes:

"Randy Yates" wrote in message
...
When James said "in the final mix," I took him to essentially be asking
the question, "Would a new 24 bit 96 kHz storage and delivery medium be
a good thing?" I.e., should we replace CDs with some new 24/96 format?

We already have Blu-Ray audio at higher bit rates than CD. They haven't
"replaced" CD yet of course, because most people realise it is an answer to
a problem they don't really have.

My feeling exactly! I believe 24/96 was also an option on the older
DVD-Audio format.
--
Randy Yates
Digital Signal Labs
http://www.digitalsignallabs.com

[Randy Yates[_2_]]

"Trevor" writes:

"geoff" wrote in message
...
Trevor wrote:
"Randy Yates" wrote in message
...
When James said "in the final mix," I took him to essentially be
asking the question, "Would a new 24 bit 96 kHz storage and delivery
medium be a good thing?" I.e., should we replace CDs with some new
24/96 format?

We already have Blu-Ray audio

... and DVD for years with that capability.

Right and SACD, which at least proves the point that you can fool some of
the people all the time, but not all the people all of the time. :-)

Right, and DVD-Audio (as I just posted in a twin thread).
--
Randy Yates
Digital Signal Labs
http://www.digitalsignallabs.com

[William Sommerwerck]

All of the transient timing information is contained in the
bandwidth.

If "bandwidth" includes phase, then I agree. Flat response does not
automatically equate to zero phase shift.

I'm surprised no one has included all-pass phase equalization in digital
controllers. I suspect that adding leading phase shift above 1kHz (or so)
would significantly improve the sound.

Unfortunately, I have no way to test this.

[William Sommerwerck]

Sample rate essentially *is* bit rate, in linear PCM...

No, it isn't.

Right. I wrote too quickly. I meant bit depth.

[Randy Yates[_2_]]

"geoff" writes:

William Sommerwerck wrote:
To my ears, Blu-ray audio sounds noticeably better than the common
run of CD sound. However...

Blu-ray audio disks are audiophile recordings, and may use better
mics, fewsd\\er electronics, less processing, etc, than Compact Disk
recordings. At the moment, I cannot say with any objectivity that
"high resolution" recordings are definitely superior. In theory,
there's no reason for them being better, as 16/44 presumably records
everything that's audible.

There is some suggestion that the difference is only in different
mastering.

Several years back, I purchased a DVD-Audio player and disks with the
intention to check out the new format. I can't remember the exact disk,
but I distinctly remember finding this to be the case, i.e., that at
least one of the disks had changed the mixdown (e.g., brought the vocals
forward, used some different reverb, or somesuch)! NOT apples-to-apples!
And I was NOT happy about it musically either - it distinctly changed
the sound of that disk, and I wanted the "original."
--
Randy Yates
Digital Signal Labs
http://www.digitalsignallabs.com

[Randy Yates[_2_]]

"William Sommerwerck" writes:

All of the transient timing information is contained in the
bandwidth.

If "bandwidth" includes phase, then I agree. Flat response does not
automatically equate to zero phase shift.

I'm surprised no one has included all-pass phase equalization in digital
controllers. I suspect that adding leading phase shift above 1kHz (or so)
would significantly improve the sound.

William,

The "hard" filtering is done digitally these days (thanks to fast
converters which greatly oversample at the analog end), and such
filtering is done with linear-phase FIRs, so there is no significant
phase shift. This is been true since the early 90s or so.
--
Randy Yates
Digital Signal Labs
http://www.digitalsignallabs.com

[Randy Yates[_2_]]

(Scott Dorsey) writes:

James T wrote:
Of course transients will correspond to ability to reproduce
equivalent frequencies. I understand that. But I was wondering to what
extent that type of information would be likely to exceed ye olde
20-20khz spec that is regarded as the baseline for human perception.

Not at all. All of the transient timing information is contained in the
bandwidth.

People in the audiophile community keep bringing up the subject of timing
as if somehow it's independent of bandwidth and if a higher sample rate
could improve timing accuracy through some mechanism other than higher
bandwidth.

I agree with your thoughs, Scott.

Now.... some argument has been made that the higher channel bandwidth
might be a good thing in that it could allow us to hear more accurate
timing cues. However, nobody who has looked for such a thing in actual
listening tests have ever found it. Doesn't mean it doesn't exist, but
nobody has found it.

As we've known (thanks to a lot of the pioneering work by Lipschitz,
Wannamaker, etc.), quantizing properly using dither (whether from analog
to digital or from high-resolution digital to lower-resolution digital)
is equivalent to adding flat, wideband noise to the original analog
signal.

So I would turn the question around: does adding a very low level of
flat noise to a signal disturb its imaging, timing, positional, or other
properties under investigating here?

My suspicion is, "not at all."
--
Randy Yates
Digital Signal Labs
http://www.digitalsignallabs.com

[William Sommerwerck]

We already have Blu-ray audio at higher bit rates than CD.
They haven't "replaced" CD yet of course, because most
people realise it is an answer to a problem they don't really
have.

That could be because they don't know how good a recording can be.

Though I much prefer CD to LP, I've never been fully happy with the "sound"
of CDs. Few of them come remotely close to the sound of my own live
recordings -- including those made on cassette decks!

SACD & BD recordings -- especially surround/multi-ch -- come much closer to
what I expect a good (ie, realistic) recording to sound like.

[Scott Dorsey]

Don Pearce wrote:
On 17 Jun 2012 08:38:14 -0400, (Scott Dorsey) wrote:
William Sommerwerck wrote:
Sample rate essentially *is* bit rate, in linear PCM...

No, it isn't.

Though quantization is a form of sampling, it is not generally referred to
that way. When people say "sample rate", they're talking about temporal
sampling.

BIT RATE (bits/sec) = SAMPLE RATE (samples/sec) * WORD LENGTH (bits/sample)

If there's one thing I learned in engineering school, it's to always do
the dimensional analysis even when you think you know what the units are.

You forgot the number of channels, Scott. Dimensionless unfortunately,
so dimensional analysis doesn't warn you.

Good point!
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Randy Yates[_2_]]

Randy Yates writes:

(Scott Dorsey) writes:

James T wrote:
Of course transients will correspond to ability to reproduce
equivalent frequencies. I understand that. But I was wondering to what
extent that type of information would be likely to exceed ye olde
20-20khz spec that is regarded as the baseline for human perception.

Not at all. All of the transient timing information is contained in the
bandwidth.

People in the audiophile community keep bringing up the subject of timing
as if somehow it's independent of bandwidth and if a higher sample rate
could improve timing accuracy through some mechanism other than higher
bandwidth.

I agree with your thoughs, Scott.

Whoa! I was still thinking on the "bits" side - which is not the
question here. Sorry!
--
Randy Yates
Digital Signal Labs
http://www.digitalsignallabs.com

[Randy Yates[_2_]]

(Scott Dorsey) writes:

Don Pearce wrote:
On 17 Jun 2012 08:38:14 -0400, (Scott Dorsey) wrote:
William Sommerwerck wrote:
Sample rate essentially *is* bit rate, in linear PCM...

No, it isn't.

Though quantization is a form of sampling, it is not generally referred to
that way. When people say "sample rate", they're talking about temporal
sampling.

BIT RATE (bits/sec) = SAMPLE RATE (samples/sec) * WORD LENGTH (bits/sample)

If there's one thing I learned in engineering school, it's to always do
the dimensional analysis even when you think you know what the units are.

You forgot the number of channels, Scott. Dimensionless unfortunately,
so dimensional analysis doesn't warn you.

Good point!

Well, you could consider SAMPLE RATE as having an implied "per channel" dimension.
--
Randy Yates
Digital Signal Labs
http://www.digitalsignallabs.com

[Scott Dorsey]

Randy Yates wrote:
So I would turn the question around: does adding a very low level of
flat noise to a signal disturb its imaging, timing, positional, or other
properties under investigating here?

My suspicion is, "not at all."

I don't know. BUT, my experience is that, using the Prism AD-124 converter,
when I change the noise shaping of the dither, I can hear changes in the
tonality of the music. All that should be changing is the frequency
distribution of the noise floor, but what I perceive is a greater change.
It's possible I have discovered a flaw in the converter and that there are
side effects unrelated to the noise floor, too. I never looked into it in
great detail, I just leave the noise shaping switch to off.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[William Sommerwerck]

I'm surprised no one has included all-pass phase equalization
in digital controllers. I suspect that adding leading phase shift
above 1kHz (or so) would significantly improve the sound.

The "hard" filtering is done digitally these days (thanks to fast
converters which greatly oversample at the analog end), and such
filtering is done with linear-phase FIRs, so there is no significant
phase shift. This is been true since the early 90s or so.

You're assuming there is no non-constant-group-delay phase shift in other
parts of the recording/playback chain -- including the speakers. For
example, analog sound recordings suffer from significant phase shift of that
sort, but it is rarely corrected for.

By the way, I invented the FIR filter almost 40 years ago. When I got to the
Patent Office, I discovered Alan Dower Blumlein had beaten me by nearly four
decades. Interestingly, the patent correctly described the principle, but as
it was not "reduced to practice" in any practical manner, it was invalid. It
appears the patent examiners were every bit as stupid and ignorant 80 years
ago as they are now.

[Sean Conolly]

"William Sommerwerck" wrote in message
...
We already have Blu-ray audio at higher bit rates than CD.
They haven't "replaced" CD yet of course, because most
people realise it is an answer to a problem they don't really
have.

That could be because they don't know how good a recording can be.

Which leads directly to another discussion that's been had he most people
have never heard a really good recording, or really good speakers, or a
really good live performance...

Sean

[Randy Yates[_2_]]

(Scott Dorsey) writes:

Randy Yates wrote:
So I would turn the question around: does adding a very low level of
flat noise to a signal disturb its imaging, timing, positional, or other
properties under investigating here?

My suspicion is, "not at all."

I don't know. BUT, my experience is that, using the Prism AD-124 converter,
when I change the noise shaping of the dither, I can hear changes in the
tonality of the music. All that should be changing is the frequency
distribution of the noise floor, but what I perceive is a greater
change.

Well, if the spectrum of the noise is changed, I wouldn't be surprised
that a human might hear a different "tonality." I would expect that both
the signal spectrum and the noise spectrum affect the composite
perceived tonality. But you must have some damned good ears!

It's possible I have discovered a flaw in the converter and that there are
side effects unrelated to the noise floor, too. I never looked into it in
great detail, I just leave the noise shaping switch to off.

Seems a perfectly reasonable way to work.
--
Randy Yates
Digital Signal Labs
http://www.digitalsignallabs.com

[Randy Yates[_2_]]

"William Sommerwerck" writes:

I'm surprised no one has included all-pass phase equalization
in digital controllers. I suspect that adding leading phase shift
above 1kHz (or so) would significantly improve the sound.

The "hard" filtering is done digitally these days (thanks to fast
converters which greatly oversample at the analog end), and such
filtering is done with linear-phase FIRs, so there is no significant
phase shift. This is been true since the early 90s or so.

You're assuming there is no non-constant-group-delay phase shift in other
parts of the recording/playback chain -- including the speakers.

You're right. I hadn't realized the topic had changed from the how the
delivery chain bandwidth and bit depth affect the overall reproduction
chain to how signal processing can be used to compensate for
recording/reproduction chain response errors.

For example, analog sound recordings suffer from significant phase
shift of that sort, but it is rarely corrected for.

By the way, I invented the FIR filter almost 40 years ago.

That's pretty neat! What was your application?
--
Randy Yates
Digital Signal Labs
http://www.digitalsignallabs.com

[Scott Dorsey]

Randy Yates wrote:
(Scott Dorsey) writes:

Randy Yates wrote:
So I would turn the question around: does adding a very low level of
flat noise to a signal disturb its imaging, timing, positional, or other
properties under investigating here?

My suspicion is, "not at all."

I don't know. BUT, my experience is that, using the Prism AD-124 converter,
when I change the noise shaping of the dither, I can hear changes in the
tonality of the music. All that should be changing is the frequency
distribution of the noise floor, but what I perceive is a greater
change.

Well, if the spectrum of the noise is changed, I wouldn't be surprised
that a human might hear a different "tonality." I would expect that both
the signal spectrum and the noise spectrum affect the composite
perceived tonality. But you must have some damned good ears!

The noise floor that is being changed is 96 dB down... in fact it's probably
lower than the noise floor of the signal going into the converter. My
ears can't be that good. I think something else might be going on but I am
at a loss as to what it really is.
--scott


--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[geoff]

Don Pearce wrote:

But it is at least correct, as opposed to yours in which you divide
instead of multiply. The results are a bit different, you will
appreciate.


Um, isn't saying 44,800/24/s is pretty specific and easy, whereas 2.304Mb/s
could be any number of sample rates or bit depths that you (probably) need
additional information and a calcator to work out any one of the relevant
paramters ?

geoff

[geoff]

Scott Dorsey wrote:
Randy Yates wrote:
So I would turn the question around: does adding a very low level of
flat noise to a signal disturb its imaging, timing, positional, or
other properties under investigating here?

My suspicion is, "not at all."

I don't know. BUT, my experience is that, using the Prism AD-124
converter, when I change the noise shaping of the dither, I can hear
changes in the tonality of the music. All that should be changing is
the frequency distribution of the noise floor, but what I perceive is
a greater change. It's possible I have discovered a flaw in the
converter and that there are side effects unrelated to the noise
floor, too. I never looked into it in great detail, I just leave the
noise shaping switch to off. --scott

Maybe it is shaping the noise (!), which causes *a change* to which the mind
is attributing greater significance than was is actually happening ?

geoff

[geoff]

Scott Dorsey wrote:
The noise floor that is being changed is 96 dB down... in fact it's
probably lower than the noise floor of the signal going into the
converter. My
ears can't be that good. I think something else might be going on
but I am at a loss as to what it really is.
--scott

I was goinfg to ask in what circumstances you where hear a signal AND the
noise floor !

geoff

[William Sommerwerck]

By the way, I invented the FIR filter almost 40 years ago.

That's pretty neat! What was your application?

De-convolution to correct speaker phase and transient errors.

Very little work has been done on this, as most EQ aims to correct only
amplitude errors. The guy who developed the Apogee speakers has apparently
done some work along these lines.

[Trevor]

"William Sommerwerck" wrote in message
...
SACDs do, generally, sound different -- and "better" than CDs. WHY is
another matter.

Of course they souind different, the why is well known, they are NOT the
same.
Whether they sound "better" is purely up to the individual releases and
listener preference.

Trevor.

[William Sommerwerck]

SACDs do, generally, sound different -- and "better" than
CDs. WHY is another matter.

Of course they souind different, the why is well known...

And the "why" is...?

Whether they sound "better" is purely up to the individual
releases and listener preference.

I disagree. Though SACDs vary, they are generally closer to "the real thing"
than CD.

[Trevor]

"William Sommerwerck" wrote in message
...
To my ears, Blu-ray audio sounds noticeably better than
the common run of CD sound. However...

Blu-ray audio disks are audiophile recordings, and may use better mics,
fewsd\\er electronics, less processing, etc, than Compact Disk
recordings.
At the moment, I cannot say with any objectivity that "high resolution"
recordings are definitely superior. In theory, there's no reason
for
them being better, as 16/44 presumably records everything that's
audible.

Simple to resample those Blu-Ray recordings to 16/44 to compare them
properly. I can't hear a difference, and never met anyone who *claims*
they
can who could prove it.
You may be the exception of course, but I wouldn't bet money on it. :-)
Tracking at 24/96 is another matter of course.

It would be useful if you actually read and understood what I wrote.

It would be more useful if you weren't so arrogant as to assume I don't!
Although I admit to not knowing what you might have thought differently but
*didn't* actually write. Even more arrogant to think I should!



The CD layer of most SACDs is derived from the DSD recording, and thus
tells
us nothing about what went on prior to the recorder. Had you read what I
actually wrote -- rather than what you wanted to read -- you'd know this.

Does not contradict what I said, so perhaps YOU should read what I wrote?
And how does a standard CD tell us "what went on prior to the recorder" or
even prior to the final master for that matter?
As many others here already know, CD is fine for the *final* product, what
happens before that is another matter, just as I said.

Trevor.

[Trevor]

"geoff" wrote in message
...
To get the bit rate in linear PCM, you multiply the sample rate by the
number of bits (bit depth), and then by the number of channels.

... which is nowhere near as concise and instantly appreciable a spec as
SR/Bits/chans .

Indeed, you can get the same overall bit rate from different combinations of
SR/bit depth/channels!

Trevor.

[Don Pearce[_3_]]

On Mon, 18 Jun 2012 08:50:00 +1200, "geoff"
wrote:

Don Pearce wrote:

But it is at least correct, as opposed to yours in which you divide
instead of multiply. The results are a bit different, you will
appreciate.


Um, isn't saying 44,800/24/s is pretty specific and easy, whereas 2.304Mb/s
could be any number of sample rates or bit depths that you (probably) need
additional information and a calcator to work out any one of the relevant
paramters ?

geoff

You are still dividing instead of multiplying. It doesn't work if you
divide.

d

[Trevor]

"Scott Dorsey" wrote in message
...
In article , Trevor
wrote:
"Les Cargill" wrote in message
...
24 bit depth is good for tracking because of the additional 8 bits
headroom.

There's no converter ever made that will give 8 bits extra headroom, and
never will be even with cryogenic cooling. Of course 3 or 4 bits can still
be worthwhile.

You're looking on the wrong of the scale. You can't bring the noise floor
down, so just run the thing on kilovolt rails....


Sadly you still need a preamp to get to the kilovolt signal, and it won't
have 144dB of DNR either.

Trevor.