[Industrial One]

On Jul 7, 8:14 am, Willem wrote:
Play your sample at half speed and listen to what happens when
you reach 11.025 Khz.

I didn't notice anything, what was I supposed to hear?

D'oh, you're right. Roughly, that is. :-)
Did you check the other example (frequency of the moon) ? I made
a mistake there too, although it is not by an order of magnitude...

Must be a bad day.

Duh, you've escaped one year of incarceration of that bottle I
"putput" you in last year, remember? Must still be suffering from
some... psychological damage.

...



On Jul 5, 9:18 pm, "Earl Kiosterud" wrote:

Sampling theory tells us that it takes at least two samples per cycle, hence the 44.1 KHz
sample rate. The highest frequency that can be captured is 22.05 KHz (Nyquist
frequency);
frequencies higher than that will create alias frequencies below 22.05 For example, an
audio frequency at 30 KHz would produce an alias frequency component at 14.1 KHz (44.1 -
30). It also produces one at 44.1 + 30, but who cares? The 20KHz audio upper limit
allows
for comfortable guard band to the Nyquist frequency.

Do I understand correct: hz is one sine loop per second, I generate a
sine sweep from 0 - 20 KHz with a specified duration, when I view with
an audio application and zoom 'till individual samples are visible, I
notice that as frequency increases, the sine waves become shorter, and
gradually begin to appear more triangular as the smaller sample
interval makes a perfect, smooth sine shape impossible. Finally, when
it reaches 20 KHz (20,000 sampling rate) the waves have reached their
limit on appearing anything that resembles a sine, and is now a
perfect triangle: one sample at the bottom, one at the top, and one at
the bottom again, like /\/\/\/\/\/\/\/\/\/\. This would technically be
the maximum, but instead, as I continue scrolling, I see the waveform
look something like a private-case of sine waves. This time, a sine
block composed of triangles. What you're saying is that beyond 22.05
is a hack that simulates higher frequencies, but don't technically
exist on a digital medium, like the waveform of the sine sweep I
created?

With regard to your swept sine, as you go towards the higher frequency sines, you'll see
fewer and fewer samples per cycle, until there are barely more than two per cycle at 20 KHz.
Your audio program should not connect them with straight lines. If anything, it should show
them as a post filter (a brick-wall at 20KHz, probably) would see them. That is the
waveform of those samples with the above-Nyquist (above 22.05 KHz) frequency components
removed. It should draw a sine. Anything above 10 KHz should be sinusoidal, since any
other function (waveshape) would need harmonics, which would fall above the 20 KHz point,
and could not appear. For example, there ain't no such thing as a 15KHz triangle, sawtooth,
square etc., wave in audio. If there were, we'd only hear the fundamental, and that, by
definition, is a single frequency component, thus a sine.

http://i28.tinypic.com/11so6lu.png (note the milliseconds = current hz
since the wave is exactly 22.05 seconds)

No such thing as 15Khz triangle? I beg to differ. But that wasn't the
point, I was asking if anything above 22.05 was possible to reproduce
digitally, you said no, but frequencies above that can still be
simulated with hacks (combining different frequencies, 14+30=44 KHz,
or... some **** like that) so I asked if that was why (see picture)
the waveform at 20 KHz looked like many units of triangle waves that
form one unit of a sine wave. Wait, 20 KHz... 15+10, the size of that
one sine wave composed of tiny triangles looked about 10 KHz... the
world makes sense again! Is this what you call "modulating 10 KHz onto
15" in order to create that alias wave?

[Arny Krueger]

wrote in message

To take a contrarians position he

Here is a good article for understanding sampling theory:
http://www.wescottdesign.com/article.../sampling.html

When you sample a signal, you have to tradeoff between
frequency response, aliasing, and ringing artifacts. For
audio I believe it's ok to have ringing since we don't
notice it.

On reproducing that signal, there is that set of
tradeoffs a second time. So you can lose frequency
response there again.

All sorts of bad things can happen, and back in the late 1970s and early
1980s they did sometimes happen.

But, its 30 years later, and digital has become very efficient and
cost-effective.

In practical systems, you aren't working with idealized
sinc filters (brickwall) so there is some dropoff in
frequency response when you sample that signal and again
when you reproduce it as sound.

In modern good digital systems, most of these problems are confined to 90%
of the Nyquist frequency.

So depending on what
analog filters cost, etc. etc. there might be some sense
in going with 96khz systems.

Never happened in the past 30 years. Remember that the Nyquist frequency of
96 KHz is 48 KHz. All we really need to do is have good performance up to
16 KHz, 20 KHz at the most. At no time in the past 30 years has good digital
audio been so bad that you needed 100% safety margin for acceptable
performance.

It definitely does make
sense to sample at 96khz at acquisition... the
oversampling is beneficial (if you sample at 48khz, you
can't get very good frequency response because the analog
filters won't let you do that).

Analog filters operating at or below 48 KHz have not been part of a modern
digital audio system for at least a decade.

2- Anyways this is just speculating. The real way to
figure it out is to do a test.

Many of us have been there and done that.

Unfortunately I haven't done so myself.

That's rather evident from the tone of your post. :-(

But according to one audio engineer,
there is an audible difference.

So what? That's one guy of how many 100,000's of people doing technical work
related to audio. I'm sure there are at least a 1,000 audio engineers who
believe that they were victims of an alien abduction. So, when are you going
to make your next pilgrimage to Area 51 or Roswell, to find what you
consider to be an authoritative opinion about high sample rate audio?

[Geoff]

wrote:
To take a contrarian position he

Here is a good article for understanding sampling theory:
http://www.wescottdesign.com/article.../sampling.html

When you sample a signal, you have to tradeoff between frequency
response, aliasing, and ringing artifacts. For audio I believe it's
ok to have ringing since we don't notice it.

Ringing has not been such a factor since 20KHZ brick-wall filters went out
of favour several decades ago.


geoff

[Geoff]

Steven Sullivan wrote:
In rec.audio.tech Earl Kiosterud wrote:


DVD audio is just for marketing. No one, with the possible
exception of a few young people who can hear above 20 KHz, and many
dogs, can hear the difference between regular 44.1K 16-bit audio and
96 or 192K sampling and 24 bits -- it's been proven, though some
will tell you they can. It's something they call "resolution" for
which they have an altar, dogma and lots of ritual. They get this
dreamy look in their eyes. Challenge it, and their veins pop out
and they go on rampages. It's likely that much of the stuff you get
on DVD-audio discs is better stuff, and has been more meticulously
recorded, hence the good sound of many of them.

Actually, with the advent of DVD-A ripping software, I've found that
the stereo mixes on many of them (the rock/pop ones at least) are
just as dynamically compressed as their modern CD counterparts. And
this at 24 bits! Sheer lunacy.

Not sheer lunacy in a technical sense, one way ort the other. It's just a
production decision, like it or not.

geoff

[Allen Watson[_2_]]

In article
,
Industrial One wrote:

Most audio files on the net are recorded at a 44 KHz sampling rate,
but it's mainly referred as "frequency." Now, humans can only hear up
to 20 KHz, so why would audio be recorded at 44 KHz (twice the audible
hearing range?) Obviously, one can notice the difference if the song
was downsampled to 22, so why not coin the standard frequency at 22
KHz instead of 44, why is the number doubled? Also, just where the
hell did the number 44,100 emerge from? Why not 40,000?

Nowadays, DVD-audio songs are recorded at 96/192 KHz, is there a
point?

And if this ain't the case, why would the sampling rate be called
"frequency?"


I think a graphical explanation or demonstration of audio sampling and
reconstruction processes might help. Take a look at

http://www.aw3rd.us/hearingdigital.htm

Cheers!
- Allen

[Pete Fraser]

"geoff" wrote in message
...

Ringing has not been such a factor since 20KHZ brick-wall filters went out
of favour several decades ago.

I don't understand this.
Regular CDs have a 44.1 kHz sampling rate.
If you want a 20 kHz audio bandwidth you need a (fairly) sharp filter.
The fact that you use a mild analog filter, oversample generously,
then filter digitally before / while decimating and writing the data
to disk does not affect the ringing.

The sharp filter is done digitally, is cheaper, and is not dispersive.
The sloppy filter is in analog, and can be minimally dispersive.
However, you still have a sharp filter which will cause ringing.

What am I missing?

Pete

[Geoff]

Pete Fraser wrote:
"geoff" wrote in message
...

Ringing has not been such a factor since 20KHZ brick-wall filters
went out of favour several decades ago.

I don't understand this.
Regular CDs have a 44.1 kHz sampling rate.
If you want a 20 kHz audio bandwidth you need a (fairly) sharp filter.
The fact that you use a mild analog filter, oversample generously,
then filter digitally before / while decimating and writing the data
to disk does not affect the ringing.

The sharp filter is done digitally, is cheaper, and is not dispersive.
The sloppy filter is in analog, and can be minimally dispersive.
However, you still have a sharp filter which will cause ringing.

What am I missing?


A 20KHz brick-wall filter can have gross ringing at audio frequencies. I'm
not sure that a digital filter has ringing at all, but if it has it is moved
up waaaay higher than audio, and can then be addressed buy a kinder and
gentler analogue filter.

geoff

[Pete Fraser]

"geoff" wrote in message
news
A 20KHz brick-wall filter can have gross ringing at audio frequencies.

It will have, if it's presented with a square wave.

I'm not sure that a digital filter has ringing at all

It certainly does.

but if it has it is moved up waaaay higher than audio, and can then be
addressed buy a kinder and gentler analogue filter.

Not really.
A brick wall digital filter will have similar ringing amplitude to that
of a brick wall analog filter. The main difference is that the digital
filter's
ringing will be symmetrical, whereas the analog filter's ringing will be
asymmetrical (unless the filter designer has been very careful with
group delay correction).

[Chris Hornbeck]

On Mon, 7 Jul 2008 19:23:35 -0700, "Pete Fraser"
wrote:

I'm not sure that a digital filter has ringing at all

It certainly does.

but if it has it is moved up waaaay higher than audio, and can then be
addressed buy a kinder and gentler analogue filter.

Not really.

A brick wall digital filter will have similar ringing amplitude to that
of a brick wall analog filter. The main difference is that the digital
filter's
ringing will be symmetrical, whereas the analog filter's ringing will be
asymmetrical (unless the filter designer has been very careful with
group delay correction).

Another way to say this is that a digital filter can make things
happen *before* their stimulus. Only Wall Street insiders can do
this in the real(time) world. And only until caught.

This isn't a good newsgroup in which to bring up the subject -
folks hereabouts tend think that you're ungrateful, and react
defensively. Arf!

If you really want to wade into the deep end, you'll want to
introduce the names of the two different filter strategies, with
just enough detail to be taken seriously. But don't expect any
thanks for the enlightenment.

Folks who've had to spend way too much time correcting way too
many mistaken ideas about digital storage tend to shoot first,
pretty often. Sign of the times, I guess.

It's Bahgdad around here for the discussion of digital storage.
"You said WHAT about the Prophet?"


All the best fortune,
Chris Hornbeck

[Earl Kiosterud]

"Industrial One" wrote in message
...

No such thing as 15Khz triangle? I beg to differ. But that wasn't the
point, I was asking if anything above 22.05 was possible to reproduce
digitally, you said no, but frequencies above that can still be
simulated with hacks (combining different frequencies, 14+30=44 KHz,
or... some **** like that) so I asked if that was why (see picture)
the waveform at 20 KHz looked like many units of triangle waves that
form one unit of a sine wave. Wait, 20 KHz... 15+10, the size of that
one sine wave composed of tiny triangles looked about 10 KHz... the
world makes sense again! Is this what you call "modulating 10 KHz onto
15" in order to create that alias wave?

If you feed a 15 KHz triangle to a CD system, you'll get only a sine out of it. The
harmonics of such a wave are at 45 KHz, 60 KHz, etc, and would not pass the 20 KHz
pre-filter in the ADC. All you have left is the fundamental at 15 Khz. That would be a
sine. Remember what I said about a single frequency in the spectrum, etc. It's useful to
think in the frequency domain, applying what's known about frequency components, filters,
aliases, modulation products, bad music, etc., not just in the amplitude domain, where you
think in terms of waveforms.

I don't know what you mean by "Many units of triangles that form one unit of a sine wave."
I wouldn't get into all that triangle stuff. Learn the frequency domain fundamentals, and
you'll be rocking and rolling soon.

I notice that Audacity displays sample points connected with straight lines. I don't think
that's meaningful. Other editors show what a post-filter, presumably at 20 KHz, would show,
along with the sample points.

Your posts are a bit confusing where you combine other posts. I need an extra bourbon.
--
Earl

[Geoff]

Earl Kiosterud wrote:
I notice that Audacity displays sample points connected with straight
lines. I don't think that's meaningful. Other editors show what a
post-filter, presumably at 20 KHz, would show, along with the sample
points.


That would be very presumptive of editing software. SOundForge shows
straight lines between samples - I would be most peeved if SF made an
arbitrary decision about a filter.

geoff

[Mr.T]

"geoff" wrote in message
...
Actually, with the advent of DVD-A ripping software, I've found that
the stereo mixes on many of them (the rock/pop ones at least) are
just as dynamically compressed as their modern CD counterparts. And
this at 24 bits! Sheer lunacy.

Not sheer lunacy in a technical sense, one way ort the other. It's just a
production decision, like it or not.

Personally I'm quite happy we don't have recordings with over 100dB DNR of
the actual music material. No one would ever be able to listen to them
without an isolation room and perfect hearing. (and your hearing wouldn't
stay that way long with 110-120dB peaks)
And what would it prove in any case, simply that it can now be done?

Most recordings these days do go a bit far the other way however. :-(

MrT.

[Edmund[_2_]]

On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:

"Edmund" wrote in message


I heard about that tests and it was criticized because the music was
played over a pair of passive loudspeakers with passive filters that
where nowhere near phase linear

As a rule, speakers are nowhere near phase linear, regardless of the
implementation of the crossover.

However, similar tests have been done with transducers that have better
phase response, and same results.

Which transducers and how much better?

Furthermore, you are ignoring the fact that linear phase microphones are
only a little bit easier to find, and as a rule they are not used to
record music.

Strange do you know why?

same problem with electrostatic speakers with step up transformers .

Same problem with 99,9% (more or less) of all loudspeakers ever made.

So what?

So what? if the problem lies there, then we need to improve
the loudspeakers don't we?

So no matter how much better SACD or DVDA can be, played over such
loudspeakers all the advantages are down the drain.

Even if you were right, you're basically admitting that SACD and DVDA
have no real world application.

The word admitting suggests that I know it all and even suggests I
"know" that SACD cannot improve anything. I don't know these things.

Don't know if this story is true but I very much like to attend such a
listening test an judge for myself.

I doubt that, the tests are blind tests.

So?

Did anyone here did attend such a test and on what kind of speakers was
it played?

I can guarantee you that they weren't phase linear.

Can it be that that is why no differences where heard?

Edmund

[Arny Krueger]

"Edmund" wrote in message

On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:

"Edmund" wrote in message


I heard about that tests and it was criticized because
the music was played over a pair of passive
loudspeakers with passive filters that where nowhere
near phase linear

As a rule, speakers are nowhere near phase linear,
regardless of the implementation of the crossover.

However, similar tests have been done with transducers
that have better phase response, and same results.

Which transducers and how much better?

Headphones - full range and no crossovers.

Furthermore, you are ignoring the fact that linear phase
microphones are only a little bit easier to find, and as
a rule they are not used to record music.

Strange do you know why?

Phase linear response does not give sufficient audible advantages to offset
the difficulty of building loudspeakers that have it.

same problem with electrostatic speakers with step up
transformers .

Same problem with 99,9% (more or less) of all
loudspeakers ever made.

So what?

So what? if the problem lies there, then we need to
improve the loudspeakers don't we?

You've missed two points:

(1) is that in the current context were non phase-linear speakers are the
rule, DVD-A and SACD formats have no audible benefits.

(2) Even when listening using headphones, which have often have better phase
response than speakers, DVD-A and SACD formats still have no audible
benefits.

So no matter how much better SACD or DVDA can be,
played over such loudspeakers all the advantages are
down the drain.

Even if you were right, you're basically admitting that
SACD and DVDA have no real world application.

The word admitting suggests that I know it all and even
suggests I "know" that SACD cannot improve anything. I
don't know these things.

Don't know if this story is true but I very much like
to attend such a listening test an judge for myself.

I doubt that the tests were blind tests.

So?

Then they were invalidated by listener bias.

Did anyone here did attend such a test and on what kind
of speakers was it played?

I can guarantee you that they weren't phase linear.

Can it be that is why no differences where heard?

No.

[Arny Krueger]

"Pasi Ojala" wrote in message

On 2008-07-07,
wrote:
When you sample a signal, you have to tradeoff between
frequency response, aliasing, and ringing artifacts.
For audio I believe it's ok to have ringing since we
don't notice it.

Ringing is not an artifact, it is how a band-limited
signal behaves. It may look funny on the computer screen,
but you don't hear it because there is nothing wrong with
it in the first place.

A very relevant insight. The so-called ringing is not due to the addition of
sound, it is due to the absence of sound.

In classic ringing, high frequency components of the signal are emphasized
by a rising frequency response characteristic.

If this happens in the audible frequency range, then the timbre of the sound
is changed.

In the case of so-called ringing due to a sharp frequency cut-off, there is
no emphasis of high frequencies. Therefore, there is no logical reason to
expect a timbre change.

The nature of the problem of high frequency ringing due to a sharp frequency
cut-off is actually due to people's brains misinterpreting an oscilloscope
trace.

[Edmund[_2_]]

On Tue, 08 Jul 2008 06:12:30 -0400, Arny Krueger wrote:

"Edmund" wrote in message

On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:

"Edmund" wrote in message


I heard about that tests and it was criticized because the music was
played over a pair of passive loudspeakers with passive filters that
where nowhere near phase linear

As a rule, speakers are nowhere near phase linear, regardless of the
implementation of the crossover.

However, similar tests have been done with transducers that have
better phase response, and same results.

Which transducers and how much better?

Headphones - full range and no crossovers.

Hmm, that is exactly what I had in mind :-)

Furthermore, you are ignoring the fact that linear phase microphones
are only a little bit easier to find, and as a rule they are not used
to record music.

Strange do you know why?

Phase linear response does not give sufficient audible advantages to
offset the difficulty of building loudspeakers that have it.

For me that is no reason not to try to record music the best
way possible.

same problem with electrostatic speakers with step up transformers .

Same problem with 99,9% (more or less) of all loudspeakers ever made.

So what?

So what? if the problem lies there, then we need to improve the
loudspeakers don't we?

You've missed two points:

(1) is that in the current context were non phase-linear speakers are
the rule, DVD-A and SACD formats have no audible benefits.

Again if the current loudspeakers aren't perfect, that is no
reason to stop searching for a "perfect" way to record music!

(2) Even when listening using headphones, which have often have better
phase response than speakers, DVD-A and SACD formats still have no
audible benefits.

that's weird.

So no matter how much better SACD or DVDA can be, played over such
loudspeakers all the advantages are down the drain.

Even if you were right, you're basically admitting that SACD and DVDA
have no real world application.

The word admitting suggests that I know it all and even suggests I
"know" that SACD cannot improve anything. I don't know these things.

Don't know if this story is true but I very much like to attend such
a listening test an judge for myself.

I doubt that the tests were blind tests.

So?

Then they were invalidated by listener bias.

Did anyone here did attend such a test and on what kind of speakers
was it played?

I can guarantee you that they weren't phase linear.

Can it be that is why no differences where heard?

No.

Suppose you are right what should be improved first in order
to realize a better sound reproduction? I easily hear the
difference between an unplugged chancel and a recorded one.:-)


Edmund

[Arny Krueger]

"Edmund" wrote in message

On Tue, 08 Jul 2008 06:12:30 -0400, Arny Krueger wrote:

"Edmund" wrote in message

On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:

"Edmund" wrote in message


I heard about that tests and it was criticized
because the music was played over a pair of passive
loudspeakers with passive filters that where nowhere
near phase linear

As a rule, speakers are nowhere near phase linear,
regardless of the implementation of the crossover.

However, similar tests have been done with transducers
that have better phase response, and same results.

Which transducers and how much better?

Headphones - full range and no crossovers.

Hmm, that is exactly what I had in mind :-)

Furthermore, you are ignoring the fact that linear
phase microphones are only a little bit easier to
find, and as a rule they are not used to record music.

Strange do you know why?

Phase linear response does not give sufficient audible
advantages to offset the difficulty of building
loudspeakers that have it.

For me that is no reason not to try to record music the
best way possible.

The recording/playback process is rife with serious imperfections, but for
the past 25 years, the digital medium has not been a significant problem.

same problem with electrostatic speakers with step up
transformers .

Same problem with 99,9% (more or less) of all
loudspeakers ever made.

So what?

So what? if the problem lies there, then we need to
improve the loudspeakers don't we?

You've missed two points:

(1) is that in the current context were non phase-linear
speakers are the rule, DVD-A and SACD formats have no
audible benefits.

Again if the current loudspeakers aren't perfect, that is
no reason to stop searching for a "perfect" way to record
music!

Nothing is perfect. The weakest links are more worthy of our time and effort
than the strongest ones.

(2) Even when listening using headphones, which have
often have better phase response than speakers, DVD-A
and SACD formats still have no audible benefits.

that's weird.

Not at all. The digital mediums have not been a serious problem for over 25
years.

So no matter how much better SACD or DVDA can be,
played over such loudspeakers all the advantages are
down the drain.

Even if you were right, you're basically admitting
that SACD and DVDA have no real world application.

The word admitting suggests that I know it all and even
suggests I "know" that SACD cannot improve anything. I
don't know these things.

Don't know if this story is true but I very much like
to attend such a listening test an judge for myself.

I doubt that the tests were blind tests.

So?

Then they were invalidated by listener bias.

Did anyone here did attend such a test and on what
kind of speakers was it played?

I can guarantee you that they weren't phase linear.

Can it be that is why no differences where heard?

No.

Suppose you are right what should be improved first in
order to realize a better sound reproduction?

Venues and transducers. Rooms, speakers, and microphones.

I easily hear the difference between an unplugged chancel and a
recorded one.:-)

Having years of experiences, involving 100s of recordings I can tell you
that the worst violence is done to recorded sound by loudspeakers and
listening rooms.

In military life there is a saying: "Amateurs worry about strategy,
professionals worry about logistics". In audio, amateurs worry about
digital recording media, and professionals worry about venues and
transducers.

[Geoff]

Arny Krueger wrote:

Phase linear response does not give sufficient audible advantages to
offset the difficulty of building loudspeakers that have it.

I want a phase-linear room.

geoff

[ChrisCoaster]

On Jul 5, 11:47*pm, Industrial One wrote:
Most audio files on the net are recorded at a 44 KHz sampling rate,
but it's mainly referred as "frequency." Now, humans can only hear up
to 20 KHz, so why would audio be recorded at 44 KHz (twice the audible
hearing range?) Obviously, one can notice the difference if the song
was downsampled to 22, so why not coin the standard frequency at 22
KHz instead of 44, why is the number doubled? Also, just where the
hell did the number 44,100 emerge from? Why not 40,000?

Nowadays, DVD-audio songs are recorded at 96/192 KHz, is there a
point?

And if this ain't the case, why would the sampling rate be called
"frequency?"
___________________
Very simple. CDs have two channels of sound: Left, Right, A, B, 1, 2,
whatever you want to call them. L22kHz + R22kHz = 44kHz.

If you lower the sampling rate to 22kHz, each channel gets a freq
response up to only 11kHz. Fine for recorded speech, or archiving
78RPM & wax cylinder recordings, but quite insufficient for anything
recorded after 1950.

-CC

[ChrisCoaster]

On Jul 5, 11:47*pm, Industrial One wrote:
Most audio files on the net are recorded at a 44 KHz sampling rate,
but it's mainly referred as "frequency." Now, humans can only hear up
to 20 KHz, so why would audio be recorded at 44 KHz (twice the audible
hearing range?) Obviously, one can notice the difference if the song
was downsampled to 22, so why not coin the standard frequency at 22
KHz instead of 44, why is the number doubled? Also, just where the
hell did the number 44,100 emerge from? Why not 40,000?

Nowadays, DVD-audio songs are recorded at 96/192 KHz, is there a
point?

And if this ain't the case, why would the sampling rate be called
"frequency?"
______________________________
Google is really screwed up tonight so I'll have to repost - sorry
Usenetters!!

The audio CD sample rate is set at 44,000Hz(44kHz) for a simple
reason: Two channels - Left, Right, A,B, whatever you want to call
them. Each channel gets up to 22kHz.

This is the second time I've tried to post this so I hope this time it
goes through.

-ChrisCoaster

[Chris Hornbeck]

On Tue, 8 Jul 2008 19:59:42 -0700 (PDT), ChrisCoaster
wrote:

And if this ain't the case, why would the sampling rate be called
"frequency?"

Very simple. CDs have two channels of sound: Left, Right, A, B, 1, 2,
whatever you want to call them. L22kHz + R22kHz = 44kHz.

If you lower the sampling rate to 22kHz, each channel gets a freq
response up to only 11kHz. Fine for recorded speech, or archiving
78RPM & wax cylinder recordings, but quite insufficient for anything
recorded after 1950.
__________________________
Google is really screwed up tonight so I'll have to repost - sorry
Usenetters!!

The audio CD sample rate is set at 44,000Hz(44kHz) for a simple
reason: Two channels - Left, Right, A,B, whatever you want to call
them. Each channel gets up to 22kHz.

This is the second time I've tried to post this so I hope this time it
goes through.

Don't use Google. It's a pathetic walking-dead simulation of Usenet.
Just say No.


Your conception of sampling is very, very mistaken. Typical of what
comes from GoogleGroups though.

All good fortune,
Chris Hornbeck

[Richard Crowley]

"ChrisCoaster" wrote ...
Very simple. CDs have two channels of sound: Left, Right, A, B, 1, 2,
whatever you want to call them. L22kHz + R22kHz = 44kHz.

Absolutely not. Each channel is sampled at 44.1KHz.
There is no "L22KHz + R22KHz = 44KHz".

[Chronic Philharmonic]

"geoff" wrote in message
...
Earl Kiosterud wrote:
I notice that Audacity displays sample points connected with straight
lines. I don't think that's meaningful. Other editors show what a
post-filter, presumably at 20 KHz, would show, along with the sample
points.


That would be very presumptive of editing software. SOundForge shows
straight lines between samples - I would be most peeved if SF made an
arbitrary decision about a filter.

If they're drawing lines between samples, they're making arbitrary decisions
about filters. They should either be properly smoothing the waveform (they
know the sample rate don't they?), or showing a histogram, in my opinion.

[Willem]

ChrisCoaster wrote:
) The audio CD sample rate is set at 44,000Hz(44kHz) for a simple
) reason: Two channels - Left, Right, A,B, whatever you want to call
) them. Each channel gets up to 22kHz.

Where/who did you get that ridiculous idea from ?


SaSW, Willem
--
Disclaimer: I am in no way responsible for any of the statements
made in the above text. For all I know I might be
drugged or something..
No I'm not paranoid. You all think I'm paranoid, don't you !
#EOT

[Mr.T]

"geoff" wrote in message
...
Phase linear response does not give sufficient audible advantages to
offset the difficulty of building loudspeakers that have it.

I want a phase-linear room.

Or even a frequency linear room!

MrT.

[Mr.T]

"Richard Crowley" wrote in message
...
ChrisCoaster" wrote ...
Very simple. CDs have two channels of sound: Left, Right, A, B, 1, 2,
whatever you want to call them. L22kHz + R22kHz = 44kHz.

Absolutely not. Each channel is sampled at 44.1KHz.
There is no "L22KHz + R22KHz = 44KHz".

You could argue that since each channel is sampled at 44.1kHz, and the data
is interleaved, then the total sample rate is "sort of" equal to 88.2kHz.
Not that *I* would though :-)

MrT.

[Pete Fraser]

"Mr.T" MrT@home wrote in message
u...

You could argue that since each channel is sampled at 44.1kHz, and the
data
is interleaved, then the total sample rate is "sort of" equal to 88.2kHz.

If I (or anyone) did, they'd be very wrong.

[Industrial One]

On Jul 8, 11:22 pm, Willem wrote:
ChrisCoaster wrote:

) The audio CD sample rate is set at 44,000Hz(44kHz) for a simple
) reason: Two channels - Left, Right, A,B, whatever you want to call
) them. Each channel gets up to 22kHz.

Where/who did you get that ridiculous idea from ?

SaSW, Willem
--
Disclaimer: I am in no way responsible for any of the statements
made in the above text. For all I know I might be
drugged or something..
No I'm not paranoid. You all think I'm paranoid, don't you !
#EOT

That's what I'd like to know. Also, that guy replied to his own post
twice. Is it just me or is Chris talking to himself? What you been
smokin' bud?

[ChrisCoaster]

On Jul 9, 10:51*am, Industrial One wrote:
On Jul 8, 11:22 pm, Willem wrote:

ChrisCoaster wrote:

) The audio CD sample rate is set at 44,000Hz(44kHz) for a simple
) reason: *Two channels - Left, Right, A,B, whatever you want to call
) them. Each channel gets up to 22kHz.

Where/who did you get that ridiculous idea from ?

SaSW, Willem
--
Disclaimer: I am in no way responsible for any of the statements
* * * * * * made in the above text. For all I know I might be
* * * * * * drugged or something..
* * * * * * No I'm not paranoid. You all think I'm paranoid, don't you !
#EOT

That's what I'd like to know. Also, that guy replied to his own post
twice. Is it just me or is Chris talking to himself? What you been
smokin' bud?
____________________
Smokin'?? I think you must have me confused with the WhiteHouse.

Go back and read my posts from late tuesday night; I did apologize to
Usenet viewers if it appeared I was double/triple posting. I use
Google Groups, not Usenet, to post to and read newsgroups, and lately
Google's been nothing short of a cluster#uck as far as how quickly it
relays my posts - if at all!

On Topic: If someone can explain the 44.x sampling rate of CDs to
someone whose math skills are limited to adding/subracting whole
numbers, it would be appreciated. My idea that 44kHZ was space enough
to contain 22kHZ for two stereo channels was obviously wrong, and
something I would like clarified/corrected after having believed it
for 20 years now.

Thanks,

-CC

[[email protected]]

On Jul 9, 4:43 pm, ChrisCoaster wrote:
The audio CD sample rate is set at 44,000Hz(44kHz) for a simple
reason: Two channels - Left, Right, A,B, whatever you want to call
them. Each channel gets up to 22kHz.

Where/who did you get that ridiculous idea from ?
On Topic: If someone can explain the 44.x sampling rate of
CDs to someone whose math skills are limited to adding/
subracting whole numbers, it would be appreciated. My
idea that 44kHZ was space enough to contain 22kHZ for
two stereo channels was obviously wrong, and something
I would like clarified/corrected after having believed it for
20 years now.

There are plenty of references on the net and in print
that have been around for at least 25 years that explained
what was going on. You never tested whether your idea
corresponded with reality until now. And it didn;t match.

Fine. You can now take what I say can compare it to,
oh IEC 60958, for example, or the original Philips/Sony
red book CD standard. Check out Pohlmann's
Principles of Digital Audio, somewhat more widely available.,

But here's the scoop. A SINGLE channel, sampled at 44.1
kHz, is, itself, for the reasons stated earlier capable of
providing a bandwidth of less than 22.05 kHz. Now, if you
want STEREO, you're going to have to have TWO channels
with that bandwidth, you will need TWICE as many samples:
on for the left, one for the right, EACH at 44.1 kHz.

In the techncial parlance used by IEC 60958, each SAMPLE
consists of a 16-bit integer sample value, and is held in a
"subframe". a left sample and a right sample together, two
subframes, constitutes a "frame", and CD audio has 44,100
frames (each holding TWO samples) per second.

The resulting data rate of CD audio, then, is

16 bits/subframe * 2 subframes/frame * 44100 frames/sec

1,411,200 bits per second. Assuming 8 bits/byte, that's
176,400 bytes/second, and given the rec book limit of
74 minutes, that's 783,216,000 bytes/CD. And CDROMs
are about 750 MB in size, oh by the way.

Now, the issue of why 44,100 samples per second gives
you a bandwidth of less than 22,050 was first demonstrated
by Nyquist around 80 years ago, and the entire principles
relating to sampling, bandwidth, and such were cast in
mathematical rigor by Shannon over 55 years ago.

But the bottom line, and the answer to your idea, is this:
you want a channel whose bandwidth is 20 kHz? Then you
HAVE to sample it at a rate GREATER THAN TWICE the
bandwidth IF you want to capture the full bandwidth with no
loss.

44,100 samples per second will do that nicely when
implemented well.

You want TWO channels at 20 kHz bandwidth? EACH channel
has to be sampled at GREATER THAN TWICE the bandiwdth.
Now you have twice as much data.

Is the sample rate now 88.2 kS/sec? No, it's TWO channels
at 44.1 kS/sec.

However, you COULD (and people HAVE) for special purposes,
us a CD to store a SINGLE channel smapled at 88.2 kS/sec
to achieve a SINGLE CHANNEL bandwidth of about 40 kHz.

No, it's not 44.1 kHz because you have two 22.05 kHz channels.
It's two independent 44.1 kHz channels.

[Geoff]

Chronic Philharmonic wrote:
"geoff" wrote in message
...
Earl Kiosterud wrote:
I notice that Audacity displays sample points connected with
straight lines. I don't think that's meaningful. Other editors
show what a post-filter, presumably at 20 KHz, would show, along
with the sample points.


That would be very presumptive of editing software. SOundForge shows
straight lines between samples - I would be most peeved if SF made an
arbitrary decision about a filter.

If they're drawing lines between samples, they're making arbitrary
decisions about filters. They should either be properly smoothing the
waveform (they know the sample rate don't they?), or showing a
histogram, in my opinion.

Naa , a line is important in case you miss a sample dot out of field of
veiw. Histogram just plain messy, and how thick would you make the columns ?

geoff

[Chris Hornbeck]

On Wed, 9 Jul 2008 13:43:17 -0700 (PDT), ChrisCoaster
wrote:

On Topic: If someone can explain the 44.x sampling rate of CDs to
someone whose math skills are limited to adding/subracting whole
numbers, it would be appreciated. My idea that 44kHZ was space enough
to contain 22kHZ for two stereo channels was obviously wrong, and
something I would like clarified/corrected after having believed it
for 20 years now.

How about this?: Each channel is sampled separately and the
sampled (and later quantized) channels are kept separate
through the whole recording and playback chain.

Discounting tiny stray leakage in the analog stages, the
two channels don't meet again until in room air.

Each channel has to meet the Nyquist criterion separately,
and (in CD format, but NOT! in MP3, for example) has no
connection to the other channel.


Maybe a good analogy would be that CD format has a sampling
rate of 44.1K samples per second with a TOTAL bit depth
(both channels) of 32 bits. And even this is more wrong
than right.

Does that make sense?

All good fortune,
Chris Hornbeck

[Richard Crowley]

"ChrisCoaster" wrote ...
On Topic: If someone can explain the 44.x sampling rate of CDs to
someone whose math skills are limited to adding/subracting whole
numbers, it would be appreciated. My idea that 44kHZ was space enough
to contain 22kHZ for two stereo channels was obviously wrong, and
something I would like clarified/corrected after having believed it
for 20 years now.

For RedBook standard CDs, the sample frequency (Fs)
is 44.1KHz by definition. The same definition also says
that audio CDs will have two channels, not more, and not
less (unfortunately). All CDs must conform to this standard
or they will not be playable on CD players.
http://en.wikipedia.org/wiki/Red_Boo...CD_standard%29

In very general terms a digital sample stream of frequency Fs
can accurately sample frequencies up to Fs/2 (1/2 of Fs)
http://en.wikipedia.org/wiki/Nyquist_rate

That means that theoretically, a 44.1KHz sample rate can
sample up to 22.05 KHz audio.

Stereo is 2 channels. That means that there are actually
Fs*2 samples per second (88.2K samples per second)

[Mr.T]

"Pete Fraser" wrote in message
...
You could argue that since each channel is sampled at 44.1kHz, and the
data
is interleaved, then the total sample rate is "sort of" equal to
88.2kHz.
Not that *I* would though :-)

If I (or anyone) did, they'd be very wrong.

But they would not necessarily be "very wrong", simply using a non standard
definition, which is all too common on these newsgroups.
Like you it seems, I prefer to stick with what is technically correct using
standard definitions though.
I do find endlessly arguing definitions and semantics tedious however.

MrT.

[Willem]

ChrisCoaster wrote:
) On Topic: If someone can explain the 44.x sampling rate of CDs to
) someone whose math skills are limited to adding/subracting whole
) numbers, it would be appreciated. My idea that 44kHZ was space enough
) to contain 22kHZ for two stereo channels was obviously wrong, and
) something I would like clarified/corrected after having believed it
) for 20 years now.

Easy:

If you have a sound with a frequency of 20kHz, that means that the wave has
to go through one complete cycle 20.000 times a second. That is: up *and*
down 20.000 times a second. To record that you need to record both the ups
and the downs, so that's 20.000 ups plus 20.000 downs, makes 40.000 samples
per second.

Why they went up from 40kHz to 44.1kHz is some weird technical reason,
explained in another part of this thread.


SaSW, Willem
--
Disclaimer: I am in no way responsible for any of the statements
made in the above text. For all I know I might be
drugged or something..
No I'm not paranoid. You all think I'm paranoid, don't you !
#EOT

[ChrisCoaster]

On Jul 10, 1:50*am, Willem wrote:
ChrisCoaster wrote:

) On Topic: *If someone can explain the 44.x sampling rate of CDs to
) someone whose math skills are limited to adding/subracting whole
) numbers, it would be appreciated. *My idea that 44kHZ was space enough
) to contain 22kHZ for two stereo channels was obviously wrong, and
) something I would like clarified/corrected after having believed it
) for 20 years now.

Easy:

If you have a sound with a frequency of 20kHz, that means that the wave has
to go through one complete cycle 20.000 times a second. *That is: up *and*
down 20.000 times a second. *To record that you need to record both the ups
and the downs, so that's 20.000 ups plus 20.000 downs, makes 40.000 samples
per second.

Why they went up from 40kHz to 44.1kHz is some weird technical reason,
explained in another part of this thread.

SaSW, Willem
--
Disclaimer: I am in no way responsible for any of the statements
* * * * * * made in the above text. For all I know I might be
* * * * * * drugged or something..
* * * * * * No I'm not paranoid. You all think I'm paranoid, don't you !
#EOT
_________________
Thanks Willem! You are the only one to finally clarify this in
ENGLISH(remember I can barely add 2+2).

As for that 44kHz bit, there are some people out there who hear like
dogs; that is they can sense sounds equal to or greater than 20kHz.
The designers of the CD standard probably chose 22kHz because a CD's
freq response actually drops off like the Cayman Wall at the ends of
it's specified range, due to its digital nature. Unlike analog, which
"rolls" off gradually at varying rates, subject to source. So they
figured, move that Cayman Wall slightly above 20kHz to ensure that
99.999999% of musical content gets on the disc. Hence, 22kHz. So now
you have the high & low portions of soundwaves up to 22kHz = 44.1(to
be exact)kHz!

-CC

[[email protected]]

On Jul 10, 7:29 am, ChrisCoaster wrote:
On Jul 10, 1:50 am, Willem wrote:

ChrisCoaster wrote:

) On Topic: If someone can explain the 44.x sampling rate of CDs to
) someone whose math skills are limited to adding/subracting whole
) numbers, it would be appreciated. My idea that 44kHZ was space enough
) to contain 22kHZ for two stereo channels was obviously wrong, and
) something I would like clarified/corrected after having believed it
) for 20 years now.

Easy:

If you have a sound with a frequency of 20kHz, that means that the wave has
to go through one complete cycle 20.000 times a second. That is: up *and*
down 20.000 times a second. To record that you need to record both the ups
and the downs, so that's 20.000 ups plus 20.000 downs, makes 40.000 samples
per second.

Why they went up from 40kHz to 44.1kHz is some weird technical reason,
explained in another part of this thread.

SaSW, Willem
--
Disclaimer: I am in no way responsible for any of the statements
made in the above text. For all I know I might be
drugged or something..
No I'm not paranoid. You all think I'm paranoid, don't you !
#EOT

_________________
Thanks Willem! You are the only one to finally clarify this in
ENGLISH

Except that the explanation, while seemingly intuitively
correct, is misleading. Among other things, it assumes
that the waveform IS sinsusoidal AND that it has "ups
and downs". It doesn't have to, and, in fact, the signal
could consist of only differeing "ups".

(remember I can barely add 2+2).

Then if you insist on casting everything into a 2nd grade
mathematical vocabulary, you're guaranteed to get it
wrong, wven when any number of people try patiently
to explain to you that a 2+2 understanding isn't going to
cut it.


As for that 44kHz bit, there are some people out there who hear like
dogs; that is they can sense sounds equal to or greater than 20kHz.
The designers of the CD standard probably chose 22kHz because a CD's
freq response actually drops off like the Cayman Wall at the ends of
it's specified range, due to its digital nature. Unlike analog, which
"rolls" off gradually at varying rates, subject to source.

Wrong. The "sampling" portion is not in the least bit
"digital", in fact, the sampler can be PURELY analog.
One very practical example is the old analog bucket-
brigade delay lines, which absolutely HAD to have
a brick-wall ANALOG anti-aliasing feature. Another
example is switched-capacitor filters: the signals
are PURELY analog, but discrete-time sampled, and
thus absolutely REQUIRE a brick-wall ANALOG filter
to prevent aliasing.

Yet another example is found in sampling oscilloscopes,
technology which comes from well before 1970. The
fignals are purely ANALOG in a discrete-time analog system.

And guess what? That brick wall ANALOG filter has
to be placed at LESS THAN TWICE THE SAMPLING
RATE to prevent aliasing. The exception is in the case
of the sampling oscilloscope which DELIBERATELY
aliases a very high frequency down to the base band.

So they
figured, move that Cayman Wall slightly above 20kHz to ensure that
99.999999% of musical content gets on the disc. Hence, 22kHz.
So now
you have the high & low portions of soundwaves up to 22kHz = 44.1(to
be exact)kHz!

Did you bother AT ALL, to read some of the posts,
including my own, which addressed QUITE SPECIFICALLY
where the 44.1 kHz sample rate came from?

They could have done 40 kHz, 45 kHz, 50 kHz. Some for the
broadcast industry even do 32 kHz sample rates.

But the ONLY reason 44.1 kHz was chosen was because
the ONLY high-speed storage and transmission medium
that was readily available at the time (late 1970's) for
digital audio was video tape recorders. And based on the
NTSC frame rate (60 Hz), the number of scan lines per frame
(525), the number of blanked lines per frame (35) leading
to 490 available lines per field or 245 per frame) and the
video bandwidth adequate for the bit rate which lead to putting
3 sample PAIR per scan line leads to:

60 * 245 * 3 = 44100 sample PAIRS per second

On 50 Hz video, a similar calculation based on 625/50 Hz
with 37 blanked lines leads to:

50 * 294 * 3 = 44100 sample PAIRS per second

You can hold on to your "Cayman cliff" understanding if
it feels right for you, but you'd be wrong. 44.1 kHz was
chosen for the reasons given here. Period. It's not a
debatable point, it's not subject to what you think you
can or cannot understand, it's a matter of established
technical and historicl fact.

You don't like that explanation? YOu can get your
arms around it? Fine, go rewrite history and change
physics while you're at it. Good luck with that.

[ChrisCoaster]

On Jul 10, 8:16*am, wrote:
On Jul 10, 7:29 am, ChrisCoaster wrote:





On Jul 10, 1:50 am, Willem wrote:

ChrisCoaster wrote:

) On Topic: *If someone can explain the 44.x sampling rate of CDs to
) someone whose math skills are limited to adding/subracting whole
) numbers, it would be appreciated. *My idea that 44kHZ was space enough
) to contain 22kHZ for two stereo channels was obviously wrong, and
) something I would like clarified/corrected after having believed it
) for 20 years now.

Easy:

If you have a sound with a frequency of 20kHz, that means that the wave has
to go through one complete cycle 20.000 times a second. *That is: up *and*
down 20.000 times a second. *To record that you need to record both the ups
and the downs, so that's 20.000 ups plus 20.000 downs, makes 40.000 samples
per second.

Why they went up from 40kHz to 44.1kHz is some weird technical reason,
explained in another part of this thread.

SaSW, Willem
--
Disclaimer: I am in no way responsible for any of the statements
* * * * * * made in the above text. For all I know I might be
* * * * * * drugged or something..
* * * * * * No I'm not paranoid. You all think I'm paranoid, don't you !
#EOT

_________________
Thanks Willem! *You are the only one to finally clarify this in
ENGLISH

Except that the explanation, while seemingly intuitively
correct, is misleading. Among other things, it assumes
that the waveform IS sinsusoidal AND that it has "ups
and downs". It doesn't have to, and, in fact, the signal
could consist of only differeing "ups".

(remember I can barely add 2+2).

Then if you insist on casting everything into a 2nd grade
mathematical vocabulary, you're guaranteed to get it
wrong, wven when any number of people try patiently
to explain to you that a 2+2 understanding isn't going to
cut it.

As for that 44kHz bit, there are some people out there who hear like
dogs; that is they can sense sounds equal to or greater than 20kHz.
The designers of the CD standard probably chose 22kHz because a CD's
freq response actually drops off like the Cayman Wall at the ends of
it's specified range, due to its digital nature. Unlike analog, which
"rolls" off gradually at varying rates, subject to source.

Wrong. The "sampling" portion is not in the least bit
"digital", in fact, the sampler can be PURELY analog.
One very practical example is the old analog bucket-
brigade delay lines, which absolutely HAD to have
a brick-wall ANALOG anti-aliasing feature. Another
example is switched-capacitor filters: the signals
are PURELY analog, but discrete-time sampled, and
thus absolutely REQUIRE a brick-wall ANALOG filter
to prevent aliasing.

Yet another example is found in sampling oscilloscopes,
technology which comes from well before 1970. The
fignals are purely ANALOG in a discrete-time analog system.

And guess what? That brick wall ANALOG filter has
to be placed at LESS THAN TWICE THE SAMPLING
RATE to prevent aliasing. The exception is in the case
of the sampling oscilloscope which DELIBERATELY
aliases a very high frequency down to the base band.

So they
figured, move that Cayman Wall slightly above 20kHz to ensure that
99.999999% of musical content gets on the disc. *Hence, 22kHz.
So now
you have the high & low portions of soundwaves up to 22kHz = 44.1(to
be exact)kHz!

Did you bother AT ALL, to read some of the posts,
including my own, which addressed QUITE SPECIFICALLY
where the 44.1 kHz sample rate came from?

They could have done 40 kHz, 45 kHz, 50 kHz. Some for the
broadcast industry even do 32 kHz sample rates.

But the ONLY reason 44.1 kHz was chosen was because
the ONLY high-speed storage and transmission medium
that was readily available at the time (late 1970's) for
digital audio was video tape recorders. And based on the
NTSC frame rate (60 Hz), the number of scan lines per frame
(525), the number of blanked lines per frame (35) leading
to 490 available lines per field or 245 per frame) *and *the
video bandwidth adequate for the bit rate which lead to putting
3 sample PAIR per scan line leads to:

* * 60 * 245 * 3 = 44100 sample PAIRS per second

On 50 Hz video, a similar calculation based on 625/50 Hz
with 37 blanked lines leads to:

* * 50 * 294 * 3 = 44100 sample PAIRS per second

You can hold on to your "Cayman cliff" understanding if
it feels right for you, but you'd be wrong. 44.1 kHz was
chosen for the reasons given here. Period. It's not a
debatable point, it's not subject to what you think you
can or cannot understand, it's a matter of established
technical and historicl fact.

You don't like that explanation? YOu can get your
arms around it? Fine, go rewrite history and change
physics while you're at it. Good luck with that.- Hide quoted text -

- Show quoted text -
_____________
Whoa, easy buddy. I can feel your heart racing through my high-speed
connection.

All I'm saying is that I suffered a significant traumatic head injury
as a child and from that point in my life was never able to keep up or
recover in math. My english/grammar on the other hand earned me not
As, not double-As, but AAA+!! Social Studies/History - fuggedaboudit
- worst grade was a B+!! It's too bad because good grammar or
history grades do not get you ahead in this world; good math & tech
skills do. We've got a guy in the WhiteHouse who can't even pronounce
NEWKYOOLER yet he's leader of the free world!

I'm not attempting to rewrite history or change anything, I just need
this explained in terms that someone who has terrible math skills can
understand!

What I meant with my Cayman wall analogy was the actual analog
frequency response curve of the CD medium. That is, if you started to
roll sine wave from 20Hz all the way up to 22kHz, and record it to a
CD, the drop off at the high end of that would be instantaneous, not a
soft gradual roll off as on a record or cassette.

[Geoff]

Willem wrote:
ChrisCoaster wrote:
) On Topic: If someone can explain the 44.x sampling rate of CDs to
) someone whose math skills are limited to adding/subracting whole
) numbers, it would be appreciated. My idea that 44kHZ was space
enough ) to contain 22kHZ for two stereo channels was obviously
wrong, and ) something I would like clarified/corrected after having
believed it ) for 20 years now.

Easy:

If you have a sound with a frequency of 20kHz, that means that the
wave has to go through one complete cycle 20.000 times a second.
That is: up *and* down 20.000 times a second. To record that you
need to record both the ups and the downs, so that's 20.000 ups plus
20.000 downs, makes 40.000 samples per second.

Why they went up from 40kHz to 44.1kHz is some weird technical reason,
explained in another part of this thread.

You are, I hope, joking. Cos that is total crap.

geoff

[ChrisCoaster]

On Jul 10, 6:23*pm, "geoff" wrote:
Willem wrote:
ChrisCoaster wrote:
) On Topic: *If someone can explain the 44.x sampling rate of CDs to
) someone whose math skills are limited to adding/subracting whole
) numbers, it would be appreciated. *My idea that 44kHZ was space
enough ) to contain 22kHZ for two stereo channels was obviously
wrong, and ) something I would like clarified/corrected after having
believed it ) for 20 years now.

Easy:

If you have a sound with a frequency of 20kHz, that means that the
wave has to go through one complete cycle 20.000 times a second.
That is: up *and* down 20.000 times a second. *To record that you
need to record both the ups and the downs, so that's 20.000 ups plus
20.000 downs, makes 40.000 samples per second.

Why they went up from 40kHz to 44.1kHz is some weird technical reason,
explained in another part of this thread.

You are, I hope, joking. *Cos that is total crap.

geoff- Hide quoted text -

- Show quoted text -
_______________
Alrighty geoff, then YOU explain it. And remember, easy on the math!

'Cause at this point I'm about hit "Wiki" in my favorites.
-CC