DVDr vs CDr.

[Golitely]

When recording on cassette tapes was still the main way for us home
audio enthusiasts (of limited budget) to record music, and CDs were
just beginning to replace the old LP, I took a cue from a friend and
made the switch to using a hi-fi video recorder. I used it to record
favorite albums, 'party tapes' and my own semi-musical meanderings,
(still kept a cassette recorder/player for some things). The reason
for doing this was of course the higher quality, audio-wise, of hi-fi
video compared to cassette tapes. Now it's a generation later and I'm
wondering if I shouldn't take the same approach with digital
recording.

With the price of stand-alone DVD recorders coming down to reasonable
levels do you think it would be a good idea to buy a DVD recorder for
my home audio system, (using it, in part, to preserve old analog
recordings), and just skip buying a CD recorder altogether? (I've got
a burner for my pc, but that's all.) What would be the pluses and
minuses of doing this? Thanks for any comments.

[Arny Krueger]

"Gareth Hardy" > wrote in
message
>> With the price of stand-alone DVD recorders coming down to reasonable
>> levels do you think it would be a good idea to buy a DVD recorder for
>> my home audio system, (using it, in part, to preserve old analog
>> recordings), and just skip buying a CD recorder altogether? (I've got
>> a burner for my pc, but that's all.) What would be the pluses and
>> minuses of doing this? Thanks for any comments.

Ever hear of using computers for audio?

It works!

> Advantages of CDR:
> Portability - you can use them in your car, at your mates etc
> Price - CDR can be obtained for about 20% the price of a DVD-R
> Readiness - Sounds like you already have the kit you need.

> Advantages of DVDR:

> Higher resolution - up to 96KHz compared to 44.1KHz for CD audio)

No audible benefits.

> More space - A CDR has just 16% the capacity of a DVDR

Good point, but in fact most practical CD-R's are full of empty space, as
actually used.

> More compression options - Dolby, MPEG1 and 2, and (of course)
> uncompressed.

Right

Disadvantages of DVD-R:

(1) DVD-R drives are at least 3-4x more expensive than CD-R drives.

(2) Recording software to actually exploit the hardware capabilities of the
DVD can be relatively expensive. We're talking $100's versus free.

(3) DVD-Rs are relatively slow recorders of CD-Rs.

[FDR]

"Arny Krueger" > wrote in message
...
> "Gareth Hardy" > wrote in
> message
> >> With the price of stand-alone DVD recorders coming down to reasonable
> >> levels do you think it would be a good idea to buy a DVD recorder for
> >> my home audio system, (using it, in part, to preserve old analog
> >> recordings), and just skip buying a CD recorder altogether? (I've got
> >> a burner for my pc, but that's all.) What would be the pluses and
> >> minuses of doing this? Thanks for any comments.
>
> Ever hear of using computers for audio?
>
> It works!
>
> > Advantages of CDR:
> > Portability - you can use them in your car, at your mates etc
> > Price - CDR can be obtained for about 20% the price of a DVD-R
> > Readiness - Sounds like you already have the kit you need.
>
> > Advantages of DVDR:
>
> > Higher resolution - up to 96KHz compared to 44.1KHz for CD audio)
>
> No audible benefits.

CD players use tricks to smooth out the waveform at high frequencies because
of the fewer sampling points there. Higher sampling rates lessen the
effects.

[Arny Krueger]

"FDR" > wrote in message

> "Arny Krueger" > wrote in message
> ...
>> "Gareth Hardy" > wrote in
>> message
>>>> With the price of stand-alone DVD recorders coming down to
>>>> reasonable levels do you think it would be a good idea to buy a
>>>> DVD recorder for my home audio system, (using it, in part, to
>>>> preserve old analog recordings), and just skip buying a CD
>>>> recorder altogether? (I've got a burner for my pc, but that's
>>>> all.) What would be the pluses and minuses of doing this? Thanks
>>>> for any comments.
>>
>> Ever hear of using computers for audio?
>>
>> It works!
>>
>>> Advantages of CDR:
>>> Portability - you can use them in your car, at your mates etc
>>> Price - CDR can be obtained for about 20% the price of a DVD-R
>>> Readiness - Sounds like you already have the kit you need.
>>
>>> Advantages of DVDR:
>>
>>> Higher resolution - up to 96KHz compared to 44.1KHz for CD audio)
>>
>> No audible benefits.

> CD players use tricks to smooth out the waveform at high frequencies
> because of the fewer sampling points there.

Irrelevant and wrong.

There are no tricks, just proven technology.

As long as there are slightly more than two sampling points, the
reconstruction of amplitude and phase is near-perfect.

Furthermore in CD audio, that happens at or above 22.05 KHz, which is way
more than people can hear the absence of in a musical/voice context.

AFAIK there are no bias-controlled, level-matched, time-synchronized
listening tests that show otherwise, and doing such tests is nearly trivial
in this day and age.

> Higher sampling rates lessen the effects.

AFAIK there are no bias-controlled, level-matched, time-synchronized
listening tests that show that either, and doing such tests is again nearly
trivial in this day and age.

If you need to hear for yourself, take your 24/96 sound card over to
www.pcabx.com and take a listen.

[Gareth Hardy]

> >>> Higher resolution - up to 96KHz compared to 44.1KHz for CD audio)
> >>
> >> No audible benefits.
>
> > CD players use tricks to smooth out the waveform at high frequencies
> > because of the fewer sampling points there.
>
> Irrelevant and wrong.
>

I'm not an audio expert, but I thought that having a higher sampling rate is
not primarily to benefit the high frequency waveforms. It creates a smoother
waveform at lower frequencies so polyphonics sound clearer.

As well as being no expert, chances are I wouldn't notice the difference
between 44.1KHz audio and 96KHz anyway.

[David]

"Gareth Hardy" > wrote in message >...
> > With the price of stand-alone DVD recorders coming down to reasonable
> > levels do you think it would be a good idea to buy a DVD recorder for
> > my home audio system, (using it, in part, to preserve old analog
> > recordings), and just skip buying a CD recorder altogether? (I've got
> > a burner for my pc, but that's all.) What would be the pluses and
> > minuses of doing this? Thanks for any comments.
>
> Advantages of CDR:
> Portability - you can use them in your car, at your mates etc
> Price - CDR can be obtained for about 20% the price of a DVD-R
> Readiness - Sounds like you already have the kit you need.
>
> Advantages of DVDR:
> Higher resolution - up to 96KHz compared to 44.1KHz for CD audio)
> More space - A CDR has just 16% the capacity of a DVDR
> More compression options - Dolby, MPEG1 and 2, and (of course) uncompressed.

A dvdr is the same as 7 cdr and you can get a good dvdr disc for less
than £1.00 now.

25 dvd-r for less than £20(remember thats the same as 175 cdr discs).

David

[FDR]

"Arny Krueger" > wrote in message
...
> "FDR" > wrote in message
>
> > "Arny Krueger" > wrote in message
> > ...
> >> "Gareth Hardy" > wrote in
> >> message
> >>>> With the price of stand-alone DVD recorders coming down to
> >>>> reasonable levels do you think it would be a good idea to buy a
> >>>> DVD recorder for my home audio system, (using it, in part, to
> >>>> preserve old analog recordings), and just skip buying a CD
> >>>> recorder altogether? (I've got a burner for my pc, but that's
> >>>> all.) What would be the pluses and minuses of doing this? Thanks
> >>>> for any comments.
> >>
> >> Ever hear of using computers for audio?
> >>
> >> It works!
> >>
> >>> Advantages of CDR:
> >>> Portability - you can use them in your car, at your mates etc
> >>> Price - CDR can be obtained for about 20% the price of a DVD-R
> >>> Readiness - Sounds like you already have the kit you need.
> >>
> >>> Advantages of DVDR:
> >>
> >>> Higher resolution - up to 96KHz compared to 44.1KHz for CD audio)
> >>
> >> No audible benefits.
>
> > CD players use tricks to smooth out the waveform at high frequencies
> > because of the fewer sampling points there.
>
> Irrelevant and wrong.
>
> There are no tricks, just proven technology.

Nyquist states that 2f is the sampling rate. Tell me what you get when any
sine wavel looks like when it's sampled at twice it's frequency. The actual
reconstruction can be that of a triangle, sawtooth or flatline depending on
the time it was sampled.

Fact is that there are fewer data points at the high end. You cna never
retrieve data that's not there. Yes, there are algorithms to shape the
signal and try to reconstruct it to the original but it's just harder and
less accurate.

Sampling somethin at 8x vs. 2x will always be preferred.


>
> As long as there are slightly more than two sampling points, the
> reconstruction of amplitude and phase is near-perfect.

How near perfect? Won't it be more near perfect with more data points?

This reminds me of the figure of the face on the Mars. Then when they got
higher resolution pictures you could easily tell that it wasn't a face.

>
> Furthermore in CD audio, that happens at or above 22.05 KHz, which is way
> more than people can hear the absence of in a musical/voice context.

CD players use steep filters to reject anything above 20khz so it doesn't
matter.

>
> AFAIK there are no bias-controlled, level-matched, time-synchronized
> listening tests that show otherwise, and doing such tests is nearly
trivial
> in this day and age.

Maybe, but I know that high end frequencies on a CD sound harsh to my ears.

>
> > Higher sampling rates lessen the effects.
>
> AFAIK there are no bias-controlled, level-matched, time-synchronized
> listening tests that show that either, and doing such tests is again
nearly
> trivial in this day and age.
>
> If you need to hear for yourself, take your 24/96 sound card over to
> www.pcabx.com and take a listen.
>
>

[Sasa [Sason] Miocic]

"Arny Krueger" > wrote in message
...
> > CD players use tricks to smooth out the waveform at high frequencies
> > because of the fewer sampling points there.
>
> Irrelevant and wrong.
>
> There are no tricks, just proven technology.

What technology? Assigning voltage to a time point?

> As long as there are slightly more than two sampling points, the
> reconstruction of amplitude and phase is near-perfect.

Yes, near perfect, or maybe harsh? Tell me how are you gonna discribe sine
wave with 2 sample points? You can't, it is gonna be a triangle wave. So
higher the freqs on CD, sharper they become. Thats why more sample points
are needed in high-end audio. And if you have good speakers, you can really,
maybe not actually hear the difference, but feel the difference between
44.1kHz and 96kHz sampling rate. I would discribe it as a more open,
brilliant, sparkling, fresh full airy sound. You can laugh but it's true :)

> Furthermore in CD audio, that happens at or above 22.05 KHz, which is way
> more than people can hear the absence of in a musical/voice context.

True that we cannot "hear" above 22k or below 20Hz. But have you ever "felt"
15Hz? Psicho acoustics are playing with that kind of problems.

[dgk]

On 25 Jun 2003 01:41:20 -0700, (Golitely) wrote:

>When recording on cassette tapes was still the main way for us home
>audio enthusiasts (of limited budget) to record music, and CDs were
>just beginning to replace the old LP, I took a cue from a friend and
>made the switch to using a hi-fi video recorder. I used it to record
>favorite albums, 'party tapes' and my own semi-musical meanderings,
>(still kept a cassette recorder/player for some things). The reason
>for doing this was of course the higher quality, audio-wise, of hi-fi
>video compared to cassette tapes. Now it's a generation later and I'm
>wondering if I shouldn't take the same approach with digital
>recording.
>
>With the price of stand-alone DVD recorders coming down to reasonable
>levels do you think it would be a good idea to buy a DVD recorder for
>my home audio system, (using it, in part, to preserve old analog
>recordings), and just skip buying a CD recorder altogether? (I've got
>a burner for my pc, but that's all.) What would be the pluses and
>minuses of doing this? Thanks for any comments.

Computer CD burners are pushing 52X. DVD burners have been stuck at
4X. This may or may not be important to you. I've been holding off on
DVD because I burn a lot of stuff (mostly live concerts) and when I
get the chance to burn some stuff that is accumulating on the hard
drive I want to burn it fast.

I'm not sure of the speed of standalone CD burners, since I can't
conceive of using one rather than a PC one that costs 1/4 as much and
does more. However it has to be much faster than DVD.

[David]

>
> Disadvantages of DVD-R:
>
> (1) DVD-R drives are at least 3-4x more expensive than CD-R drives.
>

A DVD-RW drive can be brought for less than £100 now which is only £50
more than the highest spec CD-RW drive.

> (2) Recording software to actually exploit the hardware capabilities of the
> DVD can be relatively expensive. We're talking $100's versus free.

Only about £30 for ulead dvd moviefactory or something silimar.

>
> (3) DVD-Rs are relatively slow recorders of CD-Rs.

DVD-R writes at 1/2/4x which equates to 9/18/36x cd speeds. The only
reason DVD-Rs take long to record is because they are 7 times bigger
than a cd.

David

[Nic]

dgk" > wrote in message
...
> On 25 Jun 2003 01:41:20 -0700, (Golitely) wrote:
>

> Computer CD burners are pushing 52X. DVD burners have been stuck at
> 4X. This may or may not be important to you. I've been holding off on
> DVD because I burn a lot of stuff (mostly live concerts) and when I
> get the chance to burn some stuff that is accumulating on the hard
> drive I want to burn it fast.
>
> I'm not sure of the speed of standalone CD burners, since I can't
> conceive of using one rather than a PC one that costs 1/4 as much and
> does more. However it has to be much faster than DVD.

AFAIK, CD and DVD speeds are different. My old DVD ROM drive which was 2x
DVD was actually equal to a 20x CD speed. I wouldn't use 52x speed anyway.
I use my 40x CD burner at a max of 24x most of the time. It is not always
wise to burn so fast as it can affect how some equipment can read the disc.

[Buckaroo]

Unbelievable tripe. WRONG.


> CD players use tricks to smooth out the waveform at high frequencies
because
> of the fewer sampling points there. Higher sampling rates lessen the
> effects.
>
>

[Jan Philips]

On Wed, 25 Jun 2003 19:40:40 +0200, "Sasa [Sason] Miocic"
> wrote:

> Tell me how are you gonna discribe sine wave with 2 sample points?

I think he said "more than 2 points". With a point at time=0, one
1/44,100 second later and 1/44,100 second after that, there is a
unique sine wave with frequency < 22,050 that fits those points.
Hopefully that will be close to the original sound.

[Buckaroo]

Unbelievable tripe. WRONG

"Gareth Hardy" > wrote in message
...
> > >>> Higher resolution - up to 96KHz compared to 44.1KHz for CD audio)
> > >>
> > >> No audible benefits.
> >
> > > CD players use tricks to smooth out the waveform at high frequencies
> > > because of the fewer sampling points there.
> >
> > Irrelevant and wrong.
> >
>
> I'm not an audio expert, but I thought that having a higher sampling rate
is
> not primarily to benefit the high frequency waveforms. It creates a
smoother
> waveform at lower frequencies so polyphonics sound clearer.
>
> As well as being no expert, chances are I wouldn't notice the difference
> between 44.1KHz audio and 96KHz anyway.
>
>
>
>

[Buckaroo]

Look at the theorem from Nyquist in mathematical form. You are spewing BS.

"Sasa [Sason] Miocic" > wrote in message
...
>
> "Arny Krueger" > wrote in message
> ...
> > > CD players use tricks to smooth out the waveform at high frequencies
> > > because of the fewer sampling points there.
> >
> > Irrelevant and wrong.
> >
> > There are no tricks, just proven technology.
>
> What technology? Assigning voltage to a time point?
>
> > As long as there are slightly more than two sampling points, the
> > reconstruction of amplitude and phase is near-perfect.
>
> Yes, near perfect, or maybe harsh? Tell me how are you gonna discribe sine
> wave with 2 sample points? You can't, it is gonna be a triangle wave. So
> higher the freqs on CD, sharper they become. Thats why more sample points
> are needed in high-end audio. And if you have good speakers, you can
really,
> maybe not actually hear the difference, but feel the difference between
> 44.1kHz and 96kHz sampling rate. I would discribe it as a more open,
> brilliant, sparkling, fresh full airy sound. You can laugh but it's true
:)
>
> > Furthermore in CD audio, that happens at or above 22.05 KHz, which is
way
> > more than people can hear the absence of in a musical/voice context.
>
> True that we cannot "hear" above 22k or below 20Hz. But have you ever
"felt"
> 15Hz? Psicho acoustics are playing with that kind of problems.
>
>

[FDR]

It's true. The output is filtered in a way to basically fill-in the missing
data points. Take a course in DSP sometime.

"Buckaroo" > wrote in message
...
> Unbelievable tripe. WRONG.
>
>
> > CD players use tricks to smooth out the waveform at high frequencies
> because
> > of the fewer sampling points there. Higher sampling rates lessen the
> > effects.
> >
> >
>
>

[Lon Stowell]

Or get a refund on one that teaches this. What data points
are allegedly missing?

FDR wrote:
> It's true. The output is filtered in a way to basically fill-in the missing
> data points. Take a course in DSP sometime.
>
> "Buckaroo" > wrote in message
> ...
>
>>Unbelievable tripe. WRONG.
>>
>>
>>
>>>CD players use tricks to smooth out the waveform at high frequencies
>>
>>because
>>
>>>of the fewer sampling points there. Higher sampling rates lessen the
>>>effects.
>>>
>>>
>>
>>
>
>

[Erik Harris]

On 25 Jun 2003 09:05:13 -0700, (David) wrote:

>A dvdr is the same as 7 cdr and you can get a good dvdr disc for less
>than £1.00 now.

And a good CD-R for less than 5 cents, if you keep an eye out for sales,
rebates, and other deals.

>25 dvd-r for less than £20(remember thats the same as 175 cdr discs).

For reliable discs that write faster than 1X? Faster than 2X? The 5 cent
CD's I mentioned above are usually approved for 48-52X speeds (and actually,
last time I bought a spindle, I paid 2 cents per disc, not 5).

My prices are being quoted in US Dollars, but even without any conversion,
it's plain to see that CD's are a hell of a lot cheaper per unit storage than
DVD's. CD-R has matured to the point where brand reliability is almost a
non-issue (and many unbranded discs are made by the more reputable
manufacturers anyhow), yet DVD-/+R hasn't matured to that point yet, and you
need to be a little more careful to get a brand of disc that "plays nice"
with your writer and with players. So pretty much any CD media is pretty
reliable, at 2-8 cents (US) for fastest-speed media, whereas a reliable 4X
DVD-/+R still costs over $1 here in the States (usually $2 or so). And with
prices being 10-50 times higher, buying in bulk is less of an option, which
further raises the per-unit price (I can afford to buy a spindle of 100 CD's,
but with the way prices are heading, and the way I use the media, buying a
spindle of 100 DVD's would be a huge waste of money).

It's _awfully_ hard to argue that DVD is cheaper than CD per unit storage,
unless you go the route of arguing for physical storage space (on your
shelf), and factor in the average usage of a disc (i.e. most CD-R's are not
filled to capacity).

--
Erik Harris
AIM: KngFuJoe http://www.eharrishome.com
Chinese-Indonesian MA Club http://www.eharrishome.com/cimac/

The above email address is obfuscated to try to prevent SPAM.
Replace each dollar sign with an "e" for the correct address.

[Lon Stowell]

I do realize you are making this up as you go along.

Wrong is wrong and all your hand waving and avoidance won't
change the fact that you simply do not have a clue what you
are talking about.

As for the ears hearing analog, you may want to check a good
physiology reference or articles in Scientific American as
to how the ear really perceives sound and how the signals
from the inner ear [past the bones and drum] are passed
to the brain. Not that it has one single solitary thing
to do with whether or not players have filters to "fill in
the missing data points", but you are wrong on the ears as
well.

FDR 0 Facts 2.



FDR wrote:
> You do realize that the ear hears analog, not digital, right?
>
> "Lon Stowell" > wrote in message
> ...
>
>>Or get a refund on one that teaches this. What data points
>>are allegedly missing?
>>
>>FDR wrote:
>>
>>>It's true. The output is filtered in a way to basically fill-in the
>>
> missing
>
>>>data points. Take a course in DSP sometime.
>>>
>>>"Buckaroo" > wrote in message
...
>>>
>>>
>>>>Unbelievable tripe. WRONG.
>>>>
>>>>
>>>>
>>>>
>>>>>CD players use tricks to smooth out the waveform at high frequencies
>>>>
>>>>because
>>>>
>>>>
>>>>>of the fewer sampling points there. Higher sampling rates lessen the
>>>>>effects.
>>>>>
>>>>>
>>>>
>>>>
>>>
>>
>
>

[FDR]

"Lon Stowell" > wrote in message
...
> I do realize you are making this up as you go along.
>
> Wrong is wrong and all your hand waving and avoidance won't
> change the fact that you simply do not have a clue what you
> are talking about.
>
> As for the ears hearing analog, you may want to check a good
> physiology reference or articles in Scientific American as
> to how the ear really perceives sound and how the signals
> from the inner ear [past the bones and drum] are passed
> to the brain. Not that it has one single solitary thing
> to do with whether or not players have filters to "fill in
> the missing data points", but you are wrong on the ears as
> well.

I've done a report on the ear for a class referencing medical books that my
wife has (she's a physician). I know how the ear works, so don't act
condescending to me. I also know that everyday sounds are analog in nature.
Sound pressure gets transduced though a microphone to an analog voltage,
then gets converted to binary and loses some info because of sampling and
other errors, get's reproduced to analog with some interpolation to recover
some lost info and amplified to a usable signal that can be put through a
speaker to recreate sound pressure.

[Gareth Hardy]

> Unbelievable tripe. WRONG

Not good enough, Buckaroo.

Your contribution to this thread seems to be telling everyone how crap they
are without offering any pointers as to why.

While I'm quite happy to be told I'm wrong (when I am, which is often), I
think it's only fair to state why. There are plenty of people other than me
who are equally misguided and can benefit from your huge wisdom.

So come on, enlighten us.

[Arny Krueger]

"Gareth Hardy" > wrote in
message
>>>>> Higher resolution - up to 96KHz compared to 44.1KHz for CD audio)
>>>>
>>>> No audible benefits.
>>
>>> CD players use tricks to smooth out the waveform at high frequencies
>>> because of the fewer sampling points there.
>>
>> Irrelevant and wrong.

> I'm not an audio expert, but I thought that having a higher sampling
> rate is not primarily to benefit the high frequency waveforms.

What higher sampling rates do is allow recording and playing back of waves
at higher frequencies. That's different than benefiting high frequency waves
that are recorded by a given sample rate.

> It creates a smoother waveform at lower frequencies so polyphonics sound
clearer.

Wrong. Since you admit you don't know what you are talking about, let me
tell you the basics.

A properly-designed and operating digital system provides essentially
perfect reproduction of all waves (subject to sample size and format
considerations) up to a frequency equal to about half the sample rate, AKA
the Nyquist frequency. For audio CDs that's about 22 KHz. For a variety of
practical reasons, Nyquist is hard to approach exactly , but a 44.1 KHz CD
can reasonably be expected to work up to about 20 KHz.

IOW on an audio CD, 20 Hz and 20 KHz are reproduced with equal accuracy.
There are more samples in the 20 Hz wave, but believe it or not that makes
no practical difference at all.

> As well as being no expert, chances are I wouldn't notice the
> difference between 44.1KHz audio and 96KHz anyway.

That comes from being human and not a bat or a dog.

[Arny Krueger]

"FDR" > wrote in message

> "Arny Krueger" > wrote in message
> ...
>> "FDR" > wrote in message
>>
>>> "Arny Krueger" > wrote in message
>>> ...
>>>> "Gareth Hardy" > wrote in
>>>> message
>>>>>> With the price of stand-alone DVD recorders coming down to
>>>>>> reasonable levels do you think it would be a good idea to buy a
>>>>>> DVD recorder for my home audio system, (using it, in part, to
>>>>>> preserve old analog recordings), and just skip buying a CD
>>>>>> recorder altogether? (I've got a burner for my pc, but that's
>>>>>> all.) What would be the pluses and minuses of doing this? Thanks
>>>>>> for any comments.
>>>>
>>>> Ever hear of using computers for audio?
>>>>
>>>> It works!
>>>>
>>>>> Advantages of CDR:
>>>>> Portability - you can use them in your car, at your mates etc
>>>>> Price - CDR can be obtained for about 20% the price of a DVD-R
>>>>> Readiness - Sounds like you already have the kit you need.
>>>>
>>>>> Advantages of DVDR:
>>>>
>>>>> Higher resolution - up to 96KHz compared to 44.1KHz for CD audio)
>>>>
>>>> No audible benefits.
>>
>>> CD players use tricks to smooth out the waveform at high frequencies
>>> because of the fewer sampling points there.
>>
>> Irrelevant and wrong.
>>
>> There are no tricks, just proven technology.
>
> Nyquist states that 2f is the sampling rate. Tell me what you get
> when any sine wavel looks like when it's sampled at twice it's
> frequency.

OK we've got another jerk-off who can't tell the difference between what I
said which was:

"As long as there are slightly more than two sampling points, the
reconstruction of amplitude and phase is near-perfect."

and

"Tell me what you get when any sine wave looks like when it's sampled at
twice it's
frequency."

At Nyquist and higher the digitized waveform is undefined. It can literally
be anything.


> The actual reconstruction can be that of a triangle,
> sawtooth or flatline depending on the time it was sampled.

Wrong, its undefined because sampling a wave at Nyquist is improper. I
suppose you never heard of anti-aliasing filters either?

The purpose of an anti-aliasing filter is to prevent the conundrum you
presented. Every good digital system has at least one.

If you want to waste your time worrying about improperly-designed systems,
be my guest, but not on my time or my post!

> Fact is that there are fewer data points at the high end.

It makes no difference, even in the real world. You need to spend some lab
time before you shoot of your mouth with tripe like this.

> You can never retrieve data that's not there.

Hence anti-aliasing filters.

> Yes, there are algorithms to
> shape the signal and try to reconstruct it to the original but it's
> just harder and less accurate.

Nope, you're grotesquely wrong. Nobody even tries.

> Sampling something at 8x vs. 2x will always be preferred.

You're using words you obviously don't know the meaning of.

>> As long as there are slightly more than two sampling points, the
>> reconstruction of amplitude and phase is near-perfect.
>
> How near perfect? Won't it be more near perfect with more data
> points?

Nope.

> This reminds me of the figure of the face on the Mars. Then when
> they got higher resolution pictures you could easily tell that it
> wasn't a face.

Irrelevant metaphor.

>> Furthermore in CD audio, that happens at or above 22.05 KHz, which
>> is way more than people can hear the absence of in a musical/voice
>> context.

> CD players use steep filters to reject anything above 20khz so it
> doesn't matter.

Actually that happens when the recording is digitized, not when it is
reconstructed in the CD player.

>> AFAIK there are no bias-controlled, level-matched, time-synchronized
>> listening tests that show otherwise, and doing such tests is nearly
trivial
>> in this day and age.

> Maybe, but I know that high end frequencies on a CD sound harsh to my
> ears.

The fact that your system is out of balance and needs some spectral
contouring is not my problem, and its not the fault of the digital recording
process.

>>> Higher sampling rates lessen the effects.
>
>> AFAIK there are no bias-controlled, level-matched, time-synchronized
>> listening tests that show that either, and doing such tests is again
nearly
>> trivial in this day and age.

>> If you need to hear for yourself, take your 24/96 sound card over to
>> www.pcabx.com and take a listen.

Needless to say, we've got another internet troll who won't actually sully
himself by coming face-to-face with the relevant facts.

Yawn!

[Arny Krueger]

"FDR" > wrote in message


> It's true. The output is filtered in a way to basically fill-in the
> missing data points. Take a course in DSP sometime.

The points aren't missing if they aren't necessary.

[Arny Krueger]

"David" > wrote in message
om

>> Disadvantages of DVD-R:

>> (1) DVD-R drives are at least 3-4x more expensive than CD-R drives.

> A DVD-RW drive can be brought for less than £100 now which is only £50
> more than the highest spec CD-RW drive.

Debating trade crap logic. So, the cheapest DVD recorder only costs a bit
more than the most expensive CD recorder. Do you realize how erroneous this
logic is or are you really that bloody stupid?

>> (2) Recording software to actually exploit the hardware capabilities
>> of the DVD can be relatively expensive. We're talking $100's versus
>> free.

> Only about £30 for ulead dvd moviefactory or something similar.

Tell me about how you can a 6-channel 24/96 audio recording or a 2-channel
24/192 recording with Ulead DVD Movie Factory.

It can't.

That's because this cheap software doesn't fully exploit the potential of
the DVD format, particularly for audio. It's for putting pictures and movies
on DVDs, not high quality sound!

Could you learn to read the specs before you waste my time with crap like
this?


>> (3) DVD-Rs are relatively slow recorders of CD-Rs.

> DVD-R writes at 1/2/4x which equates to 9/18/36x cd speeds. The only
> reason DVD-Rs take long to record is because they are 7 times bigger
> than a cd.

Last time I looked a cheap CD recorder can write at 52x. 52x > 36x. end of
story.

Look I have before me two PC's on a 2-channel KVM switch. If I take my hands
off the keyboard I put one hand on top of each one. One has a Pioneer DVR
104 DVD recorder and one has a Liteon 52x CD-R burner. They both have EZ CD
5 as well as other popular burning software on them.

Do you have any questions?

;-)

[Golitely]

"FDR" > wrote

> It's true. The output is filtered in a way to basically fill-in the missing
> data points. Take a course in DSP sometime.
>
> "Buckaroo" > wrote in message
> ...
> > Unbelievable tripe. WRONG.
> >
> >
> > > CD players use tricks to smooth out the waveform at high frequencies
> because
> > > of the fewer sampling points there. Higher sampling rates lessen the
> > > effects.

Thanks everyone for all the comments and info, and lively discussion,
so far. What about the issue of *recording* the digital information?
Wouldn't differences between sampling rates, number of bits and DACs
used have some impact on how precise or detailed the recording will
be? Also, concerning the audio portion of DVD recording/playback, are
there any unresolved issues surroundng competing formats, just in the
area of *stereo* playback, leaving off matters involving 5.1 and so
forth? Thanks again.

[Gareth Hardy]

> Wrong. Since you admit you don't know what you are talking about, let me
> tell you the basics.

Thanks, Arny. That's all I needed. The silence from buckaroo is deafening
(or is that a sampling rate problem).

I still don't really understand, but thanks for trying. My understanding now
is that if decoded properly, the same WAV file (say a tuba) will sound the
same at 32KHz, 48Khz and 96KHz because having extra samples at low frequency
will only smooth out to what the DAC would have done with fewer samples
anyway (I think).

[Arny Krueger]

"Golitely" > wrote in message
om
> "FDR" > wrote
>
>> It's true. The output is filtered in a way to basically fill-in the
>> missing data points. Take a course in DSP sometime.
>>
>> "Buckaroo" > wrote in message
>> ...
>>> Unbelievable tripe. WRONG.
>>>
>>>
>>>> CD players use tricks to smooth out the waveform at high
>>>> frequencies
>> because
>>>> of the fewer sampling points there. Higher sampling rates lessen
>>>> the effects.
>
> Thanks everyone for all the comments and info, and lively discussion,
> so far. What about the issue of *recording* the digital information?
> Wouldn't differences between sampling rates, number of bits and DACs
> used have some impact on how precise or detailed the recording will
> be?

Number of bits per sample, yes.

> Also, concerning the audio portion of DVD recording/playback, are
> there any unresolved issues surrounding competing formats, just in the
> area of *stereo* playback, leaving off matters involving 5.1 and so
> forth? Thanks again.

The interesting challenge is to make a 2-channel 24/96 or 24/192 DVD
recording with a reasonable outlay for recording software, i.e., less than
$100.

[Sasa [Sason] Miocic]

"Arny Krueger" > wrote in message
...
>
> Digital audio technology. You ought to try to learn how it works some
day!

I don't remember saying that I don't know the subject...

> >Assigning voltage to a time point?
>
> Oh spare me the expert-sounding horsecrap!

That horsecrap is basic fundation of digital recording...

> Yet another craphead who can't read. I specifically excluded that
> possibility when I said:
>
> "As long as there are slightly more than two sampling points, the
> reconstruction of amplitude and phase is near-perfect."

Yes, slightly more than two. And higher it goes, less it becomes. I just
concluded, and brought new thought.
Nice, call your fellowgroupmembers crapheads. Agression comes from fear. Yo
do not have to be affraid, sometimes people makes mistakes, even you...

> >So higher the freqs on CD, sharper they become.
>
> My boy you need to get your head out from that book where the sun shines
> not, and spend some time in the real world.

Yeah yeah...

> > Thats why more
> > sample points are needed in high-end audio.
>
> Nope, wrong.
>
> >And if you have good
> > speakers, you can really, maybe not actually hear the difference, but
> > feel the difference between
> > 44.1kHz and 96kHz sampling rate.
>
> Wrong. Look I told you were you could go and listen for yourself. Let me
> guess, you're too cheap to buy a good 24/96 sound card. You spent all your
> money on vinyl and toobs. Well, don't come belly-aching to me with your
> erroneous crap.

Listen, not everybody can afford expensive digital stuff. However, I use
Echoaudio mia 24/96 which is more than a good soundcard.
And no, I dont use vinyl. Tube as a preamp yes.
So you only gave one page, maybe you just learned everything from there?
Let's not make stupid conclusions okay?

> > I would describe it as a more open,
> > brilliant, sparkling, fresh full airy sound. You can laugh but it's
> > true :)
>
> It's false and its easy to demonstrate that its false. Just listen at the
> place I told you to go listen to.

It is false that 96kHz is better sounding than 44.1kHz?!
Man, this is going to my archive to laugh occasionally...

> > True that we cannot "hear" above 22k or below 20Hz. But have you ever
> > "felt" 15Hz? Psicho acoustics are playing with that kind of problems.
>
> I can hear a 15 KHz sine wave if its loud enough. Why would I need to
worry
> about whether I can feel it or not?

No, actually you can't hear it. You can feel it. As you can feel supersonic
sounds. I don't have time to explain it to you now, read some books...

> But again you can't read. What I said is
>
> "Furthermore in CD audio, that happens at or above 22.05 KHz, which is way
> more than people can hear the absence of in a musical/voice context.:
>
> If you sit and think about it and have average or better intelligence, you
> might eventually get what I was trying to say. If not, have an adult
explain
> it to you!

Maybe you should think of you additude first, and then try to speak. Sound
better, like 96k...

[David]

>
> And a good CD-R for less than 5 cents, if you keep an eye out for sales,
> rebates, and other deals.

Not here in the UK. Cheapest CDR i have seen is about 20p. The cost of
importing CD-R from the states would cancell out any saving I would
have made.

>
> >25 dvd-r for less than £20(remember thats the same as 175 cdr discs).
>
> For reliable discs that write faster than 1X? Faster than 2X? The 5 cent
> CD's I mentioned above are usually approved for 48-52X speeds (and actually,
> last time I bought a spindle, I paid 2 cents per disc, not 5).
>

Prices taken from www.blankshop.co.uk

Datawrite Yellow 4x DVD-R pack of 25 = £24.75
Datawrite Red 4x DVD-R pack of 25 = £19.74
Pioneer 4x DVD-R in slim jewel case = £4.00
Ice branded 4x DVD-R in packs of 25 = £19.99
Ice branded 4x DVD-R in a tub of 100 = £79.00
unbranded Ritek 4x DVD-R in pack of 25 = £31.24
unbranded Ritek 4x DVD-R in pack of 100 = £124.98
Datawrite Ritek 4x DVD-R in pack of 25 = £32.25
Memorex 4.7gb 4x DVD-R in a jewel case = £1.49
Memorex 4.7gb 4x DVD-R in pack of 10 = £13.90

Could go on but i won't.

David

[Arny Krueger]

"Gareth Hardy" > wrote in
message
>> Wrong. Since you admit you don't know what you are talking about,
>> let me tell you the basics.
>
> Thanks, Arny. That's all I needed. The silence from buckaroo is
> deafening (or is that a sampling rate problem).
>
> I still don't really understand, but thanks for trying. My
> understanding now is that if decoded properly, the same WAV file (say
> a tuba) will sound the same at 32KHz, 48Khz and 96KHz because having
> extra samples at low frequency will only smooth out to what the DAC
> would have done with fewer samples anyway (I think).

The understanding I want you to have is that a 96 KHz WAV file (say of a
tuba) will sound the same at 32KHz, 44 Khz and 96KHz, as long as the extra
high frequency extension afforded by the higher sample rate doesn't have any
audible consequences.

For example, a 96 KHz .WAV file won't sound the same if properly downsampled
to 22 KHz because chopping off all information above 11 KHz often DOES have
audible consequences.

OTOH, a 96 KHz wave file will sound the same if properly downsampled to 44.1
KHz because chopping off all information above 22 KHz generally DOESN'T have
audible consequences.

Anybody who doesn't believe me need only listen for themselves to the files
freely downloadable at http://www.pcabx.com/technical/sample_rates/index.htm
.. Be sure to use one of the DBT test coordinators freely downloadable from
http://www.pcabx.com/program/index.htm , and follow the instructions posted
at http://www.pcabx.com/training/getting_started.htm .

[Arny Krueger]

"Sasa [Sason] Miocic" > wrote in message

> "Arny Krueger" > wrote in message
> ...
>>
>> Digital audio technology. You ought to try to learn how it works
>> some day!
>
> I don't remember saying that I don't know the subject...
>
>>> Assigning voltage to a time point?
>>
>> Oh spare me the expert-sounding horsecrap!
>
> That horsecrap is basic fundation of digital recording...

No it's not.

>> Yet another craphead who can't read. I specifically excluded that
>> possibility when I said:

>> "As long as there are slightly more than two sampling points, the
>> reconstruction of amplitude and phase is near-perfect."

> Yes, slightly more than two. And higher it goes, less it becomes. I
> just concluded, and brought new thought.

Wrong, you added nothing but errors.

> Nice, call your fellowgroupmembers crapheads. Agression comes from
> fear. Yo do not have to be affraid, sometimes people makes mistakes,
> even you...

Sasa, you were pretending to be an expert and continue to do so.

>>> So higher the freqs on CD, sharper they become.

>> My boy you need to get your head out from that book where the sun
>> shines not, and spend some time in the real world.

> Yeah yeah...

>>> Thats why more
>>> sample points are needed in high-end audio.
>>
>> Nope, wrong.

>>> And if you have good
>>> speakers, you can really, maybe not actually hear the difference,
>>> but feel the difference between
>>> 44.1kHz and 96kHz sampling rate.

>> Wrong. Look I told you were you could go and listen for yourself.
>> Let me guess, you're too cheap to buy a good 24/96 sound card. You
>> spent all your money on vinyl and toobs. Well, don't come
>> belly-aching to me with your erroneous crap.

> Listen, not everybody can afford expensive digital stuff.

Listen 24/96 sound cards aren't all that expensive any more.

> However, I
> use Echoaudio mia 24/96 which is more than a good soundcard.

Agreed. The Mia is entirely suitable for your education in this matter.

Now hitch that Mia to the files and programs you can download from
www.pcabx.com and get started for some basic lessons in audio audibility.

> And no, I dont use vinyl. Tube as a preamp yes.
> So you only gave one page, maybe you just learned everything from
> there? Let's not make stupid conclusions okay?

You obviously didn't read that page.

>>> I would describe it as a more open,
>>> brilliant, sparkling, fresh full airy sound. You can laugh but it's
>>> true :)
>
>> It's false and its easy to demonstrate that its false. Just listen
>> at the place I told you to go listen to.

> It is false that 96kHz is better sounding than 44.1kHz?!

Yes.

> Man, this is going to my archive to laugh occasionally...

Come back when you have had your ears opened at www.pcabx.com .

>>> True that we cannot "hear" above 22k or below 20Hz. But have you
>>> ever "felt" 15Hz? Psicho acoustics are playing with that kind of
>>> problems.

>> I can hear a 15 KHz sine wave if its loud enough. Why would I need
>> to worry about whether I can feel it or not?

> No, actually you can't hear it. You can feel it. As you can feel
> supersonic sounds. I don't have time to explain it to you now, read
> some books...

>> But again you can't read. What I said is

>> "Furthermore in CD audio, that happens at or above 22.05 KHz, which
>> is way more than people can hear the absence of in a musical/voice
>> context.:

>> If you sit and think about it and have average or better
>> intelligence, you might eventually get what I was trying to say. If
>> not, have an adult explain it to you!

> Maybe you should think of you additude first, and then try to speak.
> Sound better, like 96k...

Sasa, you've already shown what your attitude is with your claim that
www.pcabx.com is just a single page. It's a whole freakin' web site that
would educate you well, were you to actually lower yourself to spend some
time with it.

But Sasa, you obviously know it all. Who am I to try to dispel your
erroneous thinking?

[Jan Philips]

On Thu, 26 Jun 2003 06:22:03 -0400, "Arny Krueger" >
wrote:

>> Nyquist states that 2f is the sampling rate. Tell me what you get
>> when any sine wavel looks like when it's sampled at twice it's
>> frequency.

>At Nyquist and higher the digitized waveform is undefined. It can literally
>be anything.

In the analog to digital conversion, a Fourier transform is used, and
an inverse Fourier transform for digital to analog, right?

>Hence anti-aliasing filters.

What exactly does it do (briefly)?

[FDR]

"Arny Krueger" > wrote in message
...

> Sasa, you've already shown what your attitude is with your claim that
> www.pcabx.com is just a single page. It's a whole freakin' web site that
> would educate you well, were you to actually lower yourself to spend some
> time with it.

This says it all about that PAGE:

"This Page created 9/22/1999

This Page last updated 06/02/2003

(c) Copyright 1999, 2000, 2001 Arnold B. Krueger, All rights reserved."


It's your PAGE and you even say it's a PAGE.

If you can't even remember what you wrote or what you have on the web then
what use are you?

[Jan Philips]

>The understanding I want you to have is that a 96 KHz WAV file (say of a
>tuba) will sound the same at 32KHz, 44 Khz and 96KHz, as long as the extra
>high frequency extension afforded by the higher sample rate doesn't have any
>audible consequences.
....
>OTOH, a 96 KHz wave file will sound the same if properly downsampled to 44.1
>KHz because chopping off all information above 22 KHz generally DOESN'T have
>audible consequences.

I'm wondering - why does DAT use a 68KHz sampling rate if it doesn't
do any good to go that high?

[FDR]

"Jan Philips" > wrote in message
...
> On Thu, 26 Jun 2003 06:22:03 -0400, "Arny Krueger" >
> wrote:
>
> >> Nyquist states that 2f is the sampling rate. Tell me what you get
> >> when any sine wavel looks like when it's sampled at twice it's
> >> frequency.
>
> >At Nyquist and higher the digitized waveform is undefined. It can
literally
> >be anything.
>
> In the analog to digital conversion, a Fourier transform is used, and
> an inverse Fourier transform for digital to analog, right?

No, generally a A/D samples a signal and then depending on the voltage
level, it determines what the corresponding binary would be. It's not that
complicated. The check is sometimes done against a decaying time constant
or a step voltage. The analog voltage input has limits (say 0 to 4 volts)
and the binary has limits (say 0 to 2^16).

[Jan Philips]

On Thu, 26 Jun 2003 18:36:29 GMT, "FDR"
> wrote:

>No, generally a A/D samples a signal and then depending on the voltage
>level, it determines what the corresponding binary would be.

Ok, and the Fourier transform isn't involved? I thought that it
probably was, since the process seems so similar to what a Fourier
transform and inverse Fourier transform does.

[Jan Philips]

On Thu, 26 Jun 2003 19:13:17 GMT, "FDR"
> wrote:

>Fourier transforms just transform from time domain to frequency domain.

Yes, I realized that later, only the D/A is like a Fourier transform.

>You can use Fourier to do processing, but for just digitizing an analog
>signal a simple D/A is the way to go.

And a simple D/A doesn't use Fourier transform? The FT would take the
data obtained form A/D and tell you which set of sine waves -
frequency and phase - (or any set of orthogonal functions) would
reproduce the original signal as closely as possible with the data you
have (subject to the Nyquist limit).

From what I've been told, the D/A conversion interpolates a smooth
curve through the data points, not simply a stair-step. If it doesn't
use the FT then what does it do - a cubic interpolation or something
like that?

[Arny Krueger]

"Jan Philips" > wrote in message

>> The understanding I want you to have is that a 96 KHz WAV file (say
>> of a tuba) will sound the same at 32KHz, 44 Khz and 96KHz, as long
>> as the extra high frequency extension afforded by the higher sample
>> rate doesn't have any audible consequences.
> ...
>> OTOH, a 96 KHz wave file will sound the same if properly downsampled
>> to 44.1 KHz because chopping off all information above 22 KHz
>> generally DOESN'T have audible consequences.
>
> I'm wondering - why does DAT use a 68KHz sampling rate if it doesn't
> do any good to go that high?

DAT uses 48 KHz and 44.1 Khz.

48 KHz was chosen to provide similar bandpass characteristics as 15 ips
analog tape. It was known at the time that 24 KHz was somewhat overkill, but
the difference between and 44.1 and 48 isn't really worth arguing about.

[Arny Krueger]

"Jan Philips" > wrote in message

> On Thu, 26 Jun 2003 19:13:17 GMT, "FDR"
> > wrote:
>
>> Fourier transforms just transform from time domain to frequency
>> domain.
>
> Yes, I realized that later, only the D/A is like a Fourier transform.
>
>> You can use Fourier to do processing, but for just digitizing an
>> analog signal a simple D/A is the way to go.
>
> And a simple D/A doesn't use Fourier transform?

Nor does an A/D.

> The FT would take the
> data obtained form A/D and tell you which set of sine waves -
> frequency and phase - (or any set of orthogonal functions) would
> reproduce the original signal as closely as possible with the data you
> have (subject to the Nyquist limit).

But you don't need sort of complexity that to just record, play back, and
transmit music.

> From what I've been told, the D/A conversion interpolates a smooth
> curve through the data points, not simply a stair-step.

The interpolation is generally done via a conceptually simple but
steep-sloped low-pass filter.

> If it doesn't
> use the FT then what does it do - a cubic interpolation or something
> like that?

A conceptually-simple low pass filter suffices wonderfully. The low pass
filter gets a little complex when one tries to have good filtering at and
above Nyquist, but smooth response below. Nevertheless, its now usually
mostly done in the digital domain via oversampling, and has low analog parts
count.

[Buckaroo]

I knew someone like Arny would jump in. Since I do this stuff for a living,
re-stating the obvious is torture.

"Gareth Hardy" > wrote in message
...
> > Wrong. Since you admit you don't know what you are talking about, let me
> > tell you the basics.
>
> Thanks, Arny. That's all I needed. The silence from buckaroo is deafening
> (or is that a sampling rate problem).
>
> I still don't really understand, but thanks for trying. My understanding
now
> is that if decoded properly, the same WAV file (say a tuba) will sound the
> same at 32KHz, 48Khz and 96KHz because having extra samples at low
frequency
> will only smooth out to what the DAC would have done with fewer samples
> anyway (I think).
>
>
>
>

[Arny Krueger]

"FDR" > wrote in message

> "Arny Krueger" > wrote in message
> ...
>
>> Sasa, you've already shown what your attitude is with your claim that
>> www.pcabx.com is just a single page. It's a whole freakin' web site
>> that would educate you well, were you to actually lower yourself to
>> spend some time with it.

> This says it all about that PAGE:

> "This Page created 9/22/1999

> This Page last updated 06/02/2003

> (c) Copyright 1999, 2000, 2001 Arnold B. Krueger, All rights
> reserved."

> It's your PAGE and you even say it's a PAGE.

Yes, it's the site's home page and every page on the site is linked out of
it, directly or indirectly.

You've never heard of doing things this way before?

> If you can't even remember what you wrote or what you have on the web
> then what use are you?

I guess the idea that a web site has a home page and all other pages on the
site are linked out of it is way over your head.

But, for some odd reason, that's how most of the WWW world is made. Since
your powers of observation seem to be at a low ebb, you'll have to trust me
about this...

[Buckaroo]

Arny,

I think these people are reading a lot of junk about perceptually based
music encoding and
crossing it into their conception of how PCM sampling works.

"Arny Krueger" > wrote in message
...
> "Jan Philips" > wrote in message
>
> > On Thu, 26 Jun 2003 19:13:17 GMT, "FDR"
> > > wrote:
> >
> >> Fourier transforms just transform from time domain to frequency
> >> domain.
> >
> > Yes, I realized that later, only the D/A is like a Fourier transform.
> >
> >> You can use Fourier to do processing, but for just digitizing an
> >> analog signal a simple D/A is the way to go.
> >
> > And a simple D/A doesn't use Fourier transform?
>
> Nor does an A/D.
>
> > The FT would take the
> > data obtained form A/D and tell you which set of sine waves -
> > frequency and phase - (or any set of orthogonal functions) would
> > reproduce the original signal as closely as possible with the data you
> > have (subject to the Nyquist limit).
>
> But you don't need sort of complexity that to just record, play back, and
> transmit music.
>
> > From what I've been told, the D/A conversion interpolates a smooth
> > curve through the data points, not simply a stair-step.
>
> The interpolation is generally done via a conceptually simple but
> steep-sloped low-pass filter.
>
> > If it doesn't
> > use the FT then what does it do - a cubic interpolation or something
> > like that?
>
> A conceptually-simple low pass filter suffices wonderfully. The low pass
> filter gets a little complex when one tries to have good filtering at and
> above Nyquist, but smooth response below. Nevertheless, its now usually
> mostly done in the digital domain via oversampling, and has low analog
parts
> count.
>
>

[Jan Philips]

On Thu, 26 Jun 2003 16:38:50 -0400, "Arny Krueger" >
wrote:

>A conceptually-simple low pass filter suffices wonderfully.

Can you briefly explain what a "conceptually-simple low pass filter"
is and how it gets from digital to analog? I don't know much about
electronics but I do know that a low-pass filter lets low frequencies
through. But I don't see how that goes from digital to analog.

If the original is a sine wave (under the Nyquist limit), it is
digitally sampled, and a D/A conversion is done with the
conceptually-simple low pass filter, you get back a sine wave (not
some step approximation to a sine wave), right?

[Buckaroo]

In mathematical information theory, info, as you call it, is that that
reduces uncertainty. Therefore, no information is lost
in digital sampling. You have the audiophiles erroneous belief that the
analog signal is somehow 'true' and 'complete' and the
digital signal is not. From the standpoint of mathematical information
theory, the digital signal is complete.

"FDR" > wrote in message
.. .
>
> "Lon Stowell" > wrote in message
> ...
> > I do realize you are making this up as you go along.
> >
> > Wrong is wrong and all your hand waving and avoidance won't
> > change the fact that you simply do not have a clue what you
> > are talking about.
> >
> > As for the ears hearing analog, you may want to check a good
> > physiology reference or articles in Scientific American as
> > to how the ear really perceives sound and how the signals
> > from the inner ear [past the bones and drum] are passed
> > to the brain. Not that it has one single solitary thing
> > to do with whether or not players have filters to "fill in
> > the missing data points", but you are wrong on the ears as
> > well.
>
> I've done a report on the ear for a class referencing medical books that
my
> wife has (she's a physician). I know how the ear works, so don't act
> condescending to me. I also know that everyday sounds are analog in
nature.
> Sound pressure gets transduced though a microphone to an analog voltage,
> then gets converted to binary and loses some info because of sampling and
> other errors, get's reproduced to analog with some interpolation to
recover
> some lost info and amplified to a usable signal that can be put through a
> speaker to recreate sound pressure.
>
>

[FDR]

"Arny Krueger" > wrote in message
...
> "FDR" > wrote in message
>
> > "Arny Krueger" > wrote in message
> > ...
> >
> >> Sasa, you've already shown what your attitude is with your claim that
> >> www.pcabx.com is just a single page. It's a whole freakin' web site
> >> that would educate you well, were you to actually lower yourself to
> >> spend some time with it.
>
> > This says it all about that PAGE:
>
> > "This Page created 9/22/1999
>
> > This Page last updated 06/02/2003
>
> > (c) Copyright 1999, 2000, 2001 Arnold B. Krueger, All rights
> > reserved."
>
> > It's your PAGE and you even say it's a PAGE.
>
> Yes, it's the site's home page and every page on the site is linked out of
> it, directly or indirectly.
>
> You've never heard of doing things this way before?

You call it a page, then you get ****ed at those that call it a page and not
a website. If you thought it was a website, then why don't you say so
yourself????

[Sasa [Sason] Miocic]

"Arny Krueger" > wrote in message
...
>
> > Nice, call your fellowgroupmembers crapheads. Agression comes from
> > fear. Yo do not have to be affraid, sometimes people makes mistakes,
> > even you...
>
> Sasa, you were pretending to be an expert and continue to do so.

Not really. I just stand behind my opinion as you do (and that simple
opinion is that 96k is better than 44.1k). And by doing so, I do not deserve
to be called a craphead...

> Listen 24/96 sound cards aren't all that expensive any more.

Good ones are always expensive. And just maybe I live in Croatia where 300$
sound card is stated as luxury in gonverment laws!

> > However, I
> > use Echoaudio mia 24/96 which is more than a good soundcard.
>
> Agreed. The Mia is entirely suitable for your education in this matter.
>
> Now hitch that Mia to the files and programs you can download from
> www.pcabx.com and get started for some basic lessons in audio audibility.

I have my ears, and don't need programs to test them. And concerning
lessons, I had mine more than enough...

> > It is false that 96kHz is better sounding than 44.1kHz?!
>
> Yes.
>
> > Man, this is going to my archive to laugh occasionally...
>
> Come back when you have had your ears opened at www.pcabx.com .

Now I saw FDR's post that owner of the page is actually you. No wonder you
are byting everyone who says differently. People have different opinions
about certain subjects. And in audio, sometimes you can be right or wrong.
As you believe I'm wrong, I believe you are too. So what?

> Sasa, you've already shown what your attitude is with your claim that
> www.pcabx.com is just a single page. It's a whole freakin' web site that
> would educate you well, were you to actually lower yourself to spend some
> time with it.

Do not catch me by a single word. pcabx IS a single web page/site/whatever.
I am here to learn from people and share knowledge. Not to try to convince,
by all means, somebody in his possible denial of truth as you are firmly
doing last couple of posts.

> But Sasa, you obviously know it all. Who am I to try to dispel your
> erroneous thinking?

You can try to defend your statements. Not by, as you energically did,
denying others...

Anyhow, my appologies if U felt insulted in any way. It was not intended.
And next time, please, have more patience...

[Buckaroo]

"FDR" > wrote in message
.. .
>
> "Buckaroo" > wrote in message
> ...
> > In mathematical information theory, info, as you call it, is that that
> > reduces uncertainty. Therefore, no information is lost
> > in digital sampling. You have the audiophiles erroneous belief that the
> > analog signal is somehow 'true' and 'complete' and the
> > digital signal is not. From the standpoint of mathematical information
> > theory, the digital signal is complete.
>
> The math might be right, but that doesn't mean using it in the real world
> makes sense or doesn't transfer easily.
>
> Mathematically, we know E=mc^2, but that doesn't mean that making a
nuclear
> bomb ie easy or that a theoretical bomb actually works.
>
> Maybe I'm getting off track here, but I know what my ears and my common
> sense tells me. I know that statistically, taking more data points gives
a
> higher accuracy result. I believe that data lost can never be retrieved.
> What do you have against higher sampling rates?
>
>
>

Sigh...at once you invoke both empiricism and mysticism. Can't argue with
that (laugh).

[Arny Krueger]

"Jan Philips" > wrote in message

> On Thu, 26 Jun 2003 16:38:50 -0400, "Arny Krueger" >
> wrote:
>
>> A conceptually-simple low pass filter suffices wonderfully.
>
> Can you briefly explain what a "conceptually-simple low pass filter"
> is and how it gets from digital to analog? I don't know much about
> electronics but I do know that a low-pass filter lets low frequencies
> through. But I don't see how that goes from digital to analog.

Ever hear of google?

Like Buckaroo I get tired of doing people's homework for them.

> If the original is a sine wave (under the Nyquist limit), it is
> digitally sampled, and a D/A conversion is done with the
> conceptually-simple low pass filter, you get back a sine wave (not
> some step approximation to a sine wave), right?

right.

[Gareth Hardy]

Well do everyone a favour then, and keep quiet when you notice someone is
wrong. It's selfish twice over to just state something is "tripe" and then
wait for someone else to fill in the gaps. Some of us use UseNet to try and
learn.

> I knew someone like Arny would jump in. Since I do this stuff for a
living,
> re-stating the obvious is torture.
> >
> > Thanks, Arny. That's all I needed. The silence from buckaroo is
deafening
> > (or is that a sampling rate problem).

[FDR]

"Buckaroo" > wrote in message
...
>
> "FDR" > wrote in message
> .. .
> >
> > "Buckaroo" > wrote in message
> > ...
> > > In mathematical information theory, info, as you call it, is that that
> > > reduces uncertainty. Therefore, no information is lost
> > > in digital sampling. You have the audiophiles erroneous belief that
the
> > > analog signal is somehow 'true' and 'complete' and the
> > > digital signal is not. From the standpoint of mathematical
information
> > > theory, the digital signal is complete.
> >
> > The math might be right, but that doesn't mean using it in the real
world
> > makes sense or doesn't transfer easily.
> >
> > Mathematically, we know E=mc^2, but that doesn't mean that making a
> nuclear
> > bomb ie easy or that a theoretical bomb actually works.
> >
> > Maybe I'm getting off track here, but I know what my ears and my common
> > sense tells me. I know that statistically, taking more data points
gives
> a
> > higher accuracy result. I believe that data lost can never be
retrieved.
> > What do you have against higher sampling rates?
> >
> >
> >
>
> Sigh...at once you invoke both empiricism and mysticism. Can't argue with
> that (laugh).

If common sense and life experience are mysticism, then I'm guilty. If my
engineering background is empiricism, then I'm also guilty.

Once again, what do you have against higher sampling rates?

Oh, and here's what Dallas Semiconductor has to say :

"Now moving into the mathematical realm, assume the wheel is a unit circle
with sine and cosine coordinates. If one samples at the positive and
negative peaks of the cosine values (which are 180 degrees out of phase),
then the Nyquist criteria is met and the original cosine values can be
reconstructed from the two sampled data points. Thus the Nyquist limit is
essential in reconstructing the original signal. As more and more points are
added, the ability to replicate the original signal improves."

http://www.dalsemi.com/appnotes.cfm/appnote_number/928

Wow, looky there, more points = improvement.

[Jan Philips]

On Fri, 27 Jun 2003 06:03:04 -0400, "Arny Krueger" >
wrote:

>> If the original is a sine wave (under the Nyquist limit), it is
>> digitally sampled, and a D/A conversion is done with the
>> conceptually-simple low pass filter, you get back a sine wave (not
>> some step approximation to a sine wave), right?
>
>right.

And although the Nyquist theorem applies to Fourier transforms, and
you said that the Nyquist theorem is at work here, the Fourier
transform is _not_ involved?

[Ian Hastie]

On Thu, 26 Jun 2003 13:41:17 -0700, Buckaroo wrote:

> I knew someone like Arny would jump in. Since I do this stuff for a
> living, re-stating the obvious is torture.

Then you'd be the perfect person to write up a web site on the technical
aspects of digital audio. Then whenever someone asks a question you can
point them to your web site. How does that sound?

--
Ian.

EOM

[FDR]

"Ian Hastie" > wrote in message
...
> On Thu, 26 Jun 2003 13:41:17 -0700, Buckaroo wrote:
>
> > I knew someone like Arny would jump in. Since I do this stuff for a
> > living, re-stating the obvious is torture.
>
> Then you'd be the perfect person to write up a web site on the technical
> aspects of digital audio. Then whenever someone asks a question you can
> point them to your web site. How does that sound?

It would seem obvious, but this type of person doesn't seem interested in
teaching.

>
> --
> Ian.
>
> EOM
>

[Erik Harris]

On Fri, 27 Jun 2003 10:09:11 -0400, Jan Philips
> wrote:

>On Thu, 26 Jun 2003 14:00:22 -0700, "Buckaroo"
> wrote:

>>In mathematical information theory, info, as you call it, is that that
>>reduces uncertainty. Therefore, no information is lost
>>in digital sampling.

>With some conditions. If the signal can be broken down to pure sine
>waves, none of them are above the Nyquist frequency, and the
>measurements are absolutely precise.

Doesn't the Nyquist frequency also rely on using a sinc function for
reconstruction? It's been awhile since I've worked with this stuff
academically of professionally (hence keeping out of this thread except to
call someone on the still-false claim that DVD-Rs are cheaper or even as
cheap per unit storage than CD-Rs), but I seem to recall that perfect
reconstruction of a signal sampled at or above 1/2 its max frequency required
the use of a perfect sinc function, which is not so easy to replicate in real
life (and is usually approximated). IF my memory on this is correct, then it
seems likely that while information technically isn't lost in the ADC
process, it may be lost in the DAC process. Whether or not that loss (if
present) would be sufficient to be audible to an untrained (or even trained)
ear, I dunno.

Given that Arny and "Buckaroo" are so adamant about 44.1kHz sampling being
absolutely perfect (forgetting bit depth for the moment) for frequencies up
to about 20kHz, I'm definitely interested in their thoughts as to what I'm
recalling incorrectly.

--
Erik Harris
AIM: KngFuJoe http://www.eharrishome.com
Chinese-Indonesian MA Club http://www.eharrishome.com/cimac/

The above email address is obfuscated to try to prevent SPAM.
Replace each dollar sign with an "e" for the correct address.

[Todd H.]

Erik Harris > writes:
> On Fri, 27 Jun 2003 10:09:11 -0400, Jan Philips
> > wrote:
>
> >On Thu, 26 Jun 2003 14:00:22 -0700, "Buckaroo"
> > wrote:
>
> >>In mathematical information theory, info, as you call it, is that that
> >>reduces uncertainty. Therefore, no information is lost
> >>in digital sampling.
>
> >With some conditions. If the signal can be broken down to pure sine
> >waves, none of them are above the Nyquist frequency, and the
> >measurements are absolutely precise.
>
> Doesn't the Nyquist frequency also rely on using a sinc function for
> reconstruction? It's been awhile since I've worked with this stuff
> academically of professionally (hence keeping out of this thread

You and me both.

> except to call someone on the still-false claim that DVD-Rs are
> cheaper or even as cheap per unit storage than CD-Rs), but I seem to
> recall that perfect reconstruction of a signal sampled at or above
> 1/2 its max frequency required the use of a perfect sinc function,
> which is not so easy to replicate in real life (and is usually
> approximated). IF my memory on this is correct, then it seems
> likely that while information technically isn't lost in the ADC
> process, it may be lost in the DAC process. Whether or not that
> loss (if present) would be sufficient to be audible to an untrained
> (or even trained) ear, I dunno.

Well, there's also that pesky matter of that band limiting filtering,
the required steepness of it to effectively eliminate aliasing
components, and the phase anomalies the filter presents in real life.

That (as I recall) is why practically oversampling is typically a Good
Thing--it relieved the burden of that perfect low pass filter, and
allowed it to be a) farther from the passband, b) less steep and c)
consequently able to introduce fewer phase anomalies than otherwise
possible.

> Given that Arny and "Buckaroo" are so adamant about 44.1kHz sampling
> being absolutely perfect (forgetting bit depth for the moment) for
> frequencies up to about 20kHz, I'm definitely interested in their
> thoughts as to what I'm recalling incorrectly.

Myself as well. There's a big difference bewteen the academics of the
Nyquist rate in discrete mathemtics theory, and the realities of the
componentry and actual filters one can manufacture and wrap their ears
around.


Best Regards,
--
Todd H.
http://www.toddh.net/

[Buckaroo]

You need to learn about frames of reference. The quote is correct....but
you cannot hear the improvement, which is only an
improvement in the physical frame of reference. You can find
double blind tests that prove this conclusively if you will Goggle for them.

Engineers are the worst offenders and failing to understand the slightest
thing about psychoacoustics.

[Buckaroo]

Quite true. The reason that the sample rate was set at 44.1 Ks/sec was to
allow for
filter roll off at the Nyquist at the upper limit of hearing, back when the
designs called
for analog output filters after the sample and hold. With improved digital
filtering,
oversampling, and other algorithms, it is a moot point now.

As an aside:

Most individuals in industrialized nations have severe attenuation of
hearing acuity under laboratory conditions above
ca. 16, 000 Hz. Another point to consider.

[Buckaroo]

I only teach post docs.

[FDR]

"Buckaroo" > wrote in message
...
> You need to learn about frames of reference. The quote is correct....but
> you cannot hear the improvement, which is only an
> improvement in the physical frame of reference. You can find
> double blind tests that prove this conclusively if you will Goggle for
them.

Wow, it took me a whole 5 seconds to find a website that says 96kHz is
better than 44.1 khz. Audibly better. And it's done by a mastering
engineer no-less.

http://www.audiomedia.com/archive/features/uk-0400/uk-0400-listeningtest/uk-0400-listeningtest.htm

Here is another article that sheds some light on the limits of human hearing
and how higher sampling rates can result in better sound:

http://www.ocf.berkeley.edu/~ashon/audio/Ultrasonics.htm

>
> Engineers are the worst offenders and failing to understand the slightest
> thing about psychoacoustics.

I feel so hurt.

[Sasa [Sason] Miocic]

"Buckaroo" > wrote in message
...
>
> Most individuals in industrialized nations have severe attenuation of
> hearing acuity under laboratory conditions above
> ca. 16, 000 Hz. Another point to consider.

Thats not the point of the discussion right now.

The point is that this topic became a flame, as it went from audio to totaly
mathematical/phisycal discussion. I pressume that nobody actually plugged
his/hers SACD player and heard for themselves. Or maybe recorded some stuff
and mixed in 44.1k and 96k, because of that everything of this was invented.
For the end, I can assure you, that my 96k mixes sound way much better than
pittiful 44.1k. If we already have a 96k, why the hell not use it, for
mother's sake!!!

End

[Buckaroo]

Wow. Hopeless.

The first article is voodoo. Laughable .. it would be rejected at any
meeting of scientists as opinion.

The second? Sorry, but those data say 'Harmonics visible' .. Who cares?
Percussion harmonics go to 100 kHz +

What do you not understand about science, perception, and how they connect
and relate to engineering/acoustics?

[Buckaroo]

The second article is a missive from an undergraduate student in a lower
division psychology class....give me a break.

"Buckaroo" > wrote in message
...
> Wow. Hopeless.
>
> The first article is voodoo. Laughable .. it would be rejected at any
> meeting of scientists as opinion.
>
> The second? Sorry, but those data say 'Harmonics visible' .. Who cares?
> Percussion harmonics go to 100 kHz +
>
> What do you not understand about science, perception, and how they connect
> and relate to engineering/acoustics?
>
>

[FDR]

"Sasa [Sason] Miocic" > wrote in message
...
>
> "Buckaroo" > wrote in message
> ...
> >
> > Most individuals in industrialized nations have severe attenuation of
> > hearing acuity under laboratory conditions above
> > ca. 16, 000 Hz. Another point to consider.
>
> Thats not the point of the discussion right now.
>
> The point is that this topic became a flame, as it went from audio to
totaly
> mathematical/phisycal discussion. I pressume that nobody actually plugged
> his/hers SACD player and heard for themselves. Or maybe recorded some
stuff
> and mixed in 44.1k and 96k, because of that everything of this was
invented.
> For the end, I can assure you, that my 96k mixes sound way much better
than
> pittiful 44.1k. If we already have a 96k, why the hell not use it, for
> mother's sake!!!

Because Nyquist says so!!!!!!!!!!!!!

>
> End
>
>

[Arny Krueger]

"Jan Philips" > wrote in message

> On Fri, 27 Jun 2003 06:03:04 -0400, "Arny Krueger" >
> wrote:
>
>>> If the original is a sine wave (under the Nyquist limit), it is
>>> digitally sampled, and a D/A conversion is done with the
>>> conceptually-simple low pass filter, you get back a sine wave (not
>>> some step approximation to a sine wave), right?
>>
>> right.
>
> And although the Nyquist theorem applies to Fourier transforms, and
> you said that the Nyquist theorem is at work here, the Fourier
> transform is _not_ involved?

right.

[Arny Krueger]

"FDR" > wrote in message

> "Buckaroo" > wrote in message
> ...
>> You need to learn about frames of reference. The quote is
>> correct....but you cannot hear the improvement, which is only an
>> improvement in the physical frame of reference. You can find
>> double blind tests that prove this conclusively if you will Goggle
>> for
> them.

> Wow, it took me a whole 5 seconds to find a website that says 96kHz is
> better than 44.1 khz. Audibly better. And it's done by a mastering
> engineer no-less.

>
http://www.audiomedia.com/archive/features/uk-0400/uk-0400-listeningtest/uk-
0400-listeningtest.htm


This page provides the following conclusions:

"This led to the following (preliminary) conclusions:
"1. A properly-designed 20kHz digital filter can be sonically invisible in a
96kHz sampled environment.

"2. Experience and this experiment suggests that 44.1kHz sampling digital
systems can sound much better simply by use of better digital filters. This
includes all the filters in compact disc players, A/Ds, etc. The effects of
cumulative filters must also be considered -
a situation similar to the familiar effects of group delay in successive
bandpass limited analogue circuits.

"3. 96kHz sampling systems do not sound better because of increased
bandwidth. The ear does not use information above 20kHz to evaluate sound.

IOW the site you cited says that 96 KHz is *NOT* audibly better than 44.1
KHz.


> Here is another article that sheds some light on the limits of human
> hearing and how higher sampling rates can result in better sound:

> http://www.ocf.berkeley.edu/~ashon/audio/Ultrasonics.htm

It's just an article written by an undergraduate student who was taking a
psychology course.

>> Engineers are the worst offenders and failing to understand the
>> slightest thing about psychoacoustics.

> I feel so hurt.

Listen for yourself at www.pcabx.com.

[FDR]

"Buckaroo" > wrote in message
...
> Wow. Hopeless.
>
> The first article is voodoo. Laughable .. it would be rejected at any
> meeting of scientists as opinion.

Here's the opininon of another:

"In the Handbook for Sound Engineers Steve Dove says anti-aliasing filters
"....exhibit serious frequency dependent delay and convoluted
frequency/phase characteristics... leaving mangled audio in their wake". He
also advocates sampling around 100 kHz, and says the result is a more open
and spacious sound. "

http://www.silcom.com/~aludwig/EARS.htm

But then again, what would a sound engineer know?

As for finding all these articles supporting your claim on Google, I didn't
find any.

[FDR]

"Arny Krueger" > wrote in message
...

>
> IOW the site you cited says that 96 KHz is *NOT* audibly better than 44.1
> KHz.

"My experiences with 96kHz/24-bit sampled recording have been exceptional.
The sound is more open, transparent, and dynamic than previous 44.1kHz
recordings, with a purer mid range and more apparent depth and space closer
to the analogue source."


>
>
> > Here is another article that sheds some light on the limits of human
> > hearing and how higher sampling rates can result in better sound:
>
> > http://www.ocf.berkeley.edu/~ashon/audio/Ultrasonics.htm
>
> It's just an article written by an undergraduate student who was taking a
> psychology course.

Do you dispute this in his report:

"Additional work supporting James Boyk's findings was done by John Atkinson,
editor of Stereophile magazine, the preeminent audiophile journal. In his
October 2000 editorial he describes spectral analyses of audio recordings,
all of which demonstrate more or less activity above 20 kHz. An interesting
finding he reports is that it is not just acoustic instruments that exhibit
ultrasonic activity - the electric guitar in bluegrass music, where
intentional feedback produces rampant clipping and the characteristic
electric guitar sound, also results in spectral content extending above 20
kHz. Furthermore, Atkinson noticed that even old analog recordings from the
'60s and earlier have captured this ultrasonic content."

Wow, old analog beats new 44.1 khz digital.

The report cites references that possibly higher frequencies are important
to perception. So if that's true, limiting sampling rates to 44.1 may be
discarding important perceptual information.

But hey, 44.1 is all we need right? Maybe all we need is to druve Hyundais
and eat soy food too.

>
> >> Engineers are the worst offenders and failing to understand the
> >> slightest thing about psychoacoustics.
>
> > I feel so hurt.
>
> Listen for yourself at www.pcabx.com.

Is that a page or a website?

[FDR]

Here's anpther non-scientific, but interesting article on 96 kHz including
some ABX tests:

"I know... I know... I can hear many of you saying there is absolutely NO
need for recording with a 96kHZ Sample Rate. Two weeks ago, I would have
agreed with you! I emphasize *would have* agreed with you! Let me state this
very clearly... YOU CAN INDEED HEAR THE DIFFERENCE when recording with a
96kHz Sample Rate!

I wouldn't have believed it myself if I hadn't heard the results. Bottom
line is that the highs sound more open and detailed. By the way... two other
folks here in my studio could pick the 96kHz track EVERY time in a blind
listening test (when compared with a 44.1kHz version). To hell with theory,
my EARS tell me there is a difference.

Want a real dose of Blasphemy? I compared recording at 96kHz and Sample Rate
converting down to 44.1, to simply recording at 44.1kHz. I couldn't believe
my ears! The track originally recorded at 96kHz and Sample Rate converted
down to 44.1kHz had much better sounding highs, maintaining much of the
character from recording at 96kHz.

This goes against everything that I have learned over the years... and goes
against accepted practice. So I don't make this statement lightly! You CAN
hear a difference... anyone who tells you otherwise hasn't tried recording
at 96kHz! Period. "

http://www.prorec.com/prorec/articles.nsf/files/4AE9C107C78E706886256688000FBE08

[Buckaroo]

Did you click on it? Did you listen? Audiophiles are even more mislead
than engineers.

[Buckaroo]

THAT is YOUR problem...the fact that you do not see the utter stupidity of
this as you read it.

[Buckaroo]

"...as it went from audio to totaly
> mathematical/phisycal discussion.


Absolutely NOT. I complain that you and others have not read anything about
the
results of psychoacoustical experimentation on hearing that links physics to
perception.


Hence your mysticism in what you believe you hear.

[FDR]

"Buckaroo" > wrote in message
...
> THAT is YOUR problem...the fact that you do not see the utter stupidity of
> this as you read it.

No, that is the classic repsonse of condescension and feigned superiority.
You add little else to this discussion. than to tell people they are wrong
and that you have secret knowledge that can't be revealed.

Your smug attitude will never change readers minds.

[FDR]

"Buckaroo" > wrote in message
...
> Did you click on it? Did you listen? Audiophiles are even more mislead
> than engineers.

Arny says I have to have aq $5000 audio card so I won't bother.

[Golitely]

"Buckaroo" > wrote

> RE the DVD issue. It is often perceptually based coding....compressed based
> on psychoacoustic models and thus
> not at all isomorphic with the original digital PCM signal (which is
> supposed to be included in stereo and possibly dolby surround,
> but may not be).

Oh, OK, I got 'cha... NOT! What might this mean in practical terms? I
guess you're saying that this might somewhat change the sound? I have
some things that have a fair amount of hiss in the original recording,
could you say if what you're talking about would have an impact on
recordings in which hiss is present? Thanks.

[Arny Krueger]

"FDR" > wrote in message


> "Buckaroo" > wrote in message
> ...

>> Did you click on it? Did you listen? Audiophiles are even more
>> mislead than engineers.
>
> Arny says I have to have aq $5000 audio card so I won't bother.

More lies.

[Arny Krueger]

"FDR" > wrote in message

> "Buckaroo" > wrote in message
> ...
>> THAT is YOUR problem...the fact that you do not see the utter
>> stupidity of this as you read it.
>
> No, that is the classic repsonse of condescension and feigned
> superiority. You add little else to this discussion. than to tell
> people they are wrong and that you have secret knowledge that can't
> be revealed.
>
> Your smug attitude will never change readers minds.

Highly ironic.