Frequency/Sample rate
Industrial One
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?"
Earl Kiosterud
"Industrial One" <industr...@hotmail.com> wrote in message
news:591b3f55-e2e1-408e...@z72g2000hsb.googlegroups.com...
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.
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. It ain't the extravagant bit depth and sampling rate. There are some damned
good-sounding CDs too. Even if you had a regular CD version and a DVD-audio version, and
the DVD-audio version sounded better, would you actually believe that the improvement was
because of the bit depth and sample rate? Couldn't be anything else, could it? How are
they going to sell DVD-audio discs if they let the CDs sound the same?
Hope this helps.
--
Earl
Richard Crowley
> Most audio files on the net are recorded at a 44 KHz sampling rate,
The "Red Book" convention for making audio CDs was developed
back in the early 1980s and established 44.1KHz as the sampling
rate. In order to maintain forwards and backwards compatibility,
all CDs must use that sample rate.
> but it's mainly referred as "frequency."
Any periodic occurance can be referred to as "freqency". Whether
it is something that happens every femtosecond (like light) or every
1000 years (like the century).
> Now, humans can only hear up to 20 KHz, so why would audio be
> recorded at 44 KHz (twice the audible hearing range?)
> so why not coin the standard frequency at 22
> KHz instead of 44, why is the number doubled?
The Nyquist-Shannon sampling therom tells us that you must
sample at *twice* the desired highest frequency to adequately
reproduce the original waveform. It is said that 22KHz was
selected as the top end (x2 = 44KHz) because of the state of
the art in filters back in those days.
> Obviously, one can notice the difference if the song was
> downsampled to 22,
You notice it because reducing the sampling rate to 22KHz
actually reduces the top end to 11KHz which many people
can detect.
http://en.wikipedia.org/wiki/Compact_disc
http://en.wikipedia.org/wiki/Red_Book_%28audio_CD_standard%29
http://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem
> Also, just where the hell did the number 44,100 emerge from?
> Why not 40,000?
It is said that 22KHz was selected as the top end to give some
space between the theoretical maximum "hi-fi" frequency of 20KHz
and the filter frequency (22KHz to allow room for the slope of the
filter. Modern techniques make most of the original parameters moot.
> Nowadays, DVD-audio songs are recorded at 96/192 KHz, is
> there a point?
No.
> And if this ain't the case, why would the sampling rate be called
> "frequency?"
Any periodic occurrence can be referred to as "frequency". Whether
it is something that happens every femtosecond (like light) or every
1000 years (like a new century). Most of us have to pay for electricity
and our billing cycle happens with a frequency of one month. This is
common scientific/engineering terminology. No great mystery.
Mr.T
"Richard Crowley" <rcro...@xp7rt.net> wrote in message
news:5u-dnb_XvbVZxe3V...@posted.pcez...
> Any periodic occurance can be referred to as "freqency". Whether
> it is something that happens every femtosecond (like light) or every
> 1000 years (like the century).
Only *one* century every thousand years where you live?
<Snip>
> Any periodic occurrence can be referred to as "frequency". Whether
> it is something that happens every femtosecond (like light) or every
> 1000 years (like a new century).
Obviously not a typo then.
MrT.
Edmund
> "Industrial One" <industr...@hotmail.com> wrote in message
> news:591b3f55-e2e1-408e-8b84-
d8e801...@z72g2000hsb.googlegroups.com...
>> 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?"
>
> 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.
>
> 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.
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 same problem with electrostatic
speakers with step up transformers . So no matter how much better
SACD or DVDA can be, played over such loudspeakers all the
advantages are down the drain.
Don't know if this story is true but I very much like to attend such
a listening test an judge for myself.
Did anyone here did attend such a test and on what kind of speakers
was it played?
Edmund
Chronic Philharmonic
"Industrial One" <industr...@hotmail.com> wrote in message
news:591b3f55-e2e1-408e...@z72g2000hsb.googlegroups.com...
> 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?
It was because they used video recorders for mastering prototype and first
generation CDs, and it was the nearest available frequency that was greater
than 40KHz needed to meet the sampling Nyquist requirement of at least two
samples for the highest frequency to be recorded (20KHz). See also:
http://www.cs.columbia.edu/~hgs/audio/44.1.html
> Nowadays, DVD-audio songs are recorded at 96/192 KHz, is there a
> point?
Marketing. There is no defensible mathematical requirement for it.
> And if this ain't the case, why would the sampling rate be called
> "frequency?"
The sampling rate is a frequency. CD audio is sampled at a frequency of
44.1KHz.
John Phillips
I assume this refers to the Meyer & Moran paper in JAES (see
http://www.aes.org/e-lib/browse.cfm?elib=14195). If so, I understand
it was conducted over a number of different "high-res" systems but each
test using the same 44.1 kHz/16 bit A/D - D/A loop. For example here's
a subsequent comment from E. Brad Meyer:
"... But it was a near certainty that someone in that part of the
industry would claim that with a 'real audiophile system' the
differences would have been obvious. So we found such a system and
gave its owner and his friends a chance. We conducted that test
with the same rigor as the others; levels of the two signals
were matched within 0.1 dB at 1 kHz, and then the subjects were
asked to choose their best material and listen however they usually
do, to maximize their aural acuity."
--
John Phillips
Arny Krueger
news:48708734$0$17237$bf49...@news.tele2.nl
> 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.
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.
> 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 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.
> 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.
> 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.
cr88192
"Industrial One" <industr...@hotmail.com> wrote in message
news:591b3f55-e2e1-408e...@z72g2000hsb.googlegroups.com...
> 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?
>
as others have noted...
> Nowadays, DVD-audio songs are recorded at 96/192 KHz, is there a
> point?
>
not really for listening as least, but for "audio as data", higher sampling
rate and bit depth makes sense.
of course, it is rather unlikely that end users would be doing the levels of
processing (notable doppler shifting, signal analysis, ...) that would
justify this. of course, in my case, I am usually dealing with lower-quality
input, so it works fairly well to just use 44.1kHz 16-bits anyways (one gets
a better payoff trying to write good quality filtering functions).
likewise, IMO, for MP3s much above about 128 or 192 kbps, I don't personally
hear any real difference (actually, for internet radio at least, I prefer a
little lower bitrates, as at least then the stream has less stalling and
buffering issues).
for something unrelated it is a thought that, for "audio as data" uses, one
could stuff a little bit higher-range data into the 16-bit samples, by
representing a slightly bigger range (say, 24 bits), as log-scaled values
within the 16 bit range. of course, this would likely reduce quality a
little, and add a little noise, if played back as 16 bit audio (mostly,
really quiet stuff that would normally be quantized away is left).
log10(8388608)=6.9236899, log10(32768)=4.51545
6.9236899/4.51545=1.53333333
...
actually, recently I have been suspecting that, mathematically (and for a
defined dynamic range) log-scaled values may be much more accurate for a
given number of bits than an actual floating-point format (in particular, I
think that splitting the mantissa and exponent wastes some amount of the
value range).
of course, a claim like this would require testing, and even then, who would
really care?...
it only really makes that much difference for 24 bits and lower (where
FP-style encoding starts breaking down anyways). yes, the accuracy of 16-bit
half-floats suck, not much debate here. I guess the question is if log-scale
values would be more accurate when covering the same range (2^16 to 2^-16).
it may have practical relevance though, since log-scale values are quicker
and easier to encode and decode than hfloats (given modern HW tends not to
support them anyways).
would still need to be verified though.
> And if this ain't the case, why would the sampling rate be called
> "frequency?"
as others have noted.
police also buy doughnuts with a high frequency...
doesn't mean they are chirping at the cashier...
steve
>Most audio files on the net are recorded at a 44 KHz sampling rate,
>but it's mainly referred as "frequency."
while 'frequency' is true of any periodic function
(as noted in other posts), 'sampling rate' is much
more precise. Essentially, the sample rate is the carrier
of the information in the waveform.
>Now, humans can only hear up
>to 20 KHz, so why would audio be recorded at 44 KHz (twice the audible
>hearing range?)
technically, any waveform requires at least two samples per wave.
I say 'technically', because to fully replicate the wave, with
all of its harmonics, takes many more samples than two.
at two samples, for example, you really cannot tell
whether the waveform is sine, square, sawtooth, or whatever
variant the harmonics may imbue to the waveform. Even
at four samples, the original waveform is only approximated.
(i.e. sine can look square or sawtooth depending upon
at what degrees of arc the sample is taken.)
that said, few humans can detect the difference between
a sine, square, or sawtooth above 10k. But there is
a difference. How we as humans interpret that difference
is not easily quantifiable--some may speak in terms of
clarity, crispness, 'air', 'musicality', or what have
you--not very useful terms to the engineer.
As well, sampling at a fixed frequency any other
frequency pattern will result in sampling-induced
harmonics, non-musical, that are in fact lower
than the fundamental of the wave. the amplitude
of these harmonics reduces greatly with increased
number of samples per wave, so at under, say, 5K,
they are not noticable, when sampling at 44.1k.
finally, the lower the sampling frequency, the
increased number of artifacts created when resampling.
So, if one recorded some cuts at 44.1 and others at
48k, the movement back and forth to digitally (or analog)
mix will have the effect of altering the higher-frequency
waveforms in progressive generations of resampling.
the bottom line is that sampling at 44.1k has
a noticable and significant impact on all waveforms
over 11k, and some modest impact over 5.5k.
Impacts increase with successive generations
of resampling (any time bit rate changes, and
in particular, compression.
Measuring whether humans can detect such a
change is torturously difficult.
Sampling at 96k raises the affected frequencies
to the top of the hearing range, and 192k well beyond.
to me, the greatest benefit of higher sampling rate
is in the recording and mixing stages, where the higher
rate eliminates any detectable resampling distoration
(detectable=by a scope of by a computer image of the wave).
prior to the 96 and 192 rates, i found that recording
and mixing all within 44.1 resulted in less resampling
bias and artifacts than switching between 48 and 44.1.
>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?
again, sampling rate is NOT audio frequency.
>Nowadays, DVD-audio songs are recorded at 96/192 KHz, is there a
>point?
again, recording at 96/192 may be a good idea,
but it is definitely not required for playback.
>And if this ain't the case, why would the sampling rate be called
>"frequency?"
because the frequency of a sine waveform involves a
continuous function that goes both positive and negative
during one complete 'wave'. There are generally two
zero crossings during a wave. Let's say, for a moment,
that I sampled the audio energy at each zero
crossing--exactly two samples per waveform, or at exactly
22.05 KHz. Then, upon playback, i iterpolate a
straight line between the samples. What waveform
would result?
-steve
Don Pearce
Utter cock. And of course sampling theory states that you need MORE THAN
two samples per wave, not at least two samples per wave. As long as you
have more than two, the wave is uniquely and totally described - any
further samples are unneeded and superfluous. The situation described in
the paragraph above is exactly the reason why you need more than two
samples; at exactly two the wanted and the first alias collide, and as
they are 180 degrees out of phase with each other, they cancel.
d
d
steve
>steve wrote:
>> because the frequency of a sine waveform involves a
>> continuous function that goes both positive and negative
>> during one complete 'wave'. There are generally two
>> zero crossings during a wave. Let's say, for a moment,
>> that I sampled the audio energy at each zero
>> crossing--exactly two samples per waveform, or at exactly
>> 22.05 KHz. Then, upon playback, i iterpolate a
>> straight line between the samples. What waveform
>> would result?
>>
>> -steve
>Utter cock.
ok.
>And of course sampling theory states that you need MORE THAN
>two samples per wave, not at least two samples per wave.
...so, if i require "more than two per wave", then
are you saying that the highest frequency distinguishable
from 44.1k is something less than 15k (44.1 divided by 3)?
>As long as you have more than two, the wave is uniquely
>and totally described - any further samples are unneeded
>and superfluous.
only if you assume a perfect sine wave with no
overtones. To achieve faithful replication of up to 20k,
then at least 3 samples per wave at 20k would be necessary,
true? I mean, 44.1 and 48k wouldn't cut it.
only something greater than 60k. e.g. 96k.
>The situation described in the paragraph above is exactly
>the reason why you need more than two samples; at exactly
>two the wanted and the first alias collide, and as
>they are 180 degrees out of phase with each other, they cancel.
not exactly. sure, the resultant wave would be 'nothing',
but not because the sampling rate is out of phase with the
frequency, but rather, perfectly in phase with the zero
crossings. the sound pressure level at the zero crossings
is zero (or -infinity from nominal).
So, if the sample rate is, say 22k exactly,
and the frequency rate is 11k exactly, depending upon when
the sample measures the spl at that moment, you might get
a perfect sawtooth or you might get nothing.
of course, such a situation is not likely in the 'real world',
because no natural sound source would be that perfectly
in lock step with the sampling frequency. Nevertheless,
'just two' samples per wave is insufficient to faithfully
replicate a waveform, and invites distortion.
i would agree that 'more than two' are thus necessary.
which, in my klunker way of putting it, was what i
was trying to say in the first place.
regards,
-steve
dpierce.ca...@gmail.com
> Don Pearce wrote:
> >steve wrote:
> >> because the frequency of a sine waveform involves a
> >> continuous function that goes both positive and negative
> >> during one complete 'wave'. There are generally two
> >> zero crossings during a wave. Let's say, for a moment,
> >> that I sampled the audio energy at each zero
> >> crossing--exactly two samples per waveform, or at exactly
> >> 22.05 KHz. Then, upon playback, i iterpolate a
> >> straight line between the samples. What waveform
> >> would result?
>
> >> -steve
> >Utter cock.
>
> ok.
>
> >And of course sampling theory states that you need MORE THAN
> >two samples per wave, not at least two samples per wave.
>
> ...so, if i require "more than two per wave", then
> are you saying that the highest frequency distinguishable
> from 44.1k is something less than 15k (44.1 divided by 3)?
Try 44.1k divided by 2.1, or 2.0000000001. Both are,
mathematically speaking, more than 2 and fully meet
the requirements of the Bysquist sampling theorem.
Given the first case, that would indicate that the widest
bandwidth possible if 44.1kHz/2.1 or 21 kHz. The latter
would indicate the widest bandwidth is 21.0499999... kHz.
> >As long as you have more than two, the wave is uniquely
> >and totally described - any further samples are unneeded
> >and superfluous.
>
> only if you assume a perfect sine wave with no
> overtones.
No, the ONLY assumption of Nyquist Shannon is that
ANY waveform, sinew wave, periodic, impulsive, noise-
like, ANY WAVEFORM WHATSOEVER, be band-limited
to less than 1/2 the sample rate. That's it.
> To achieve faithful replication of up to 20k,
> then at least 3 samples per wave at 20k would be necessary,
> true?
False
> I mean, 44.1 and 48k wouldn't cut it.
> only something greater than 60k. e.g. 96k.
If the bandwidth is limited to 20 kHz, and that is the
bandwidth of ANY waveform, then a sample rate greater
than twice that bandwidth is ALL that's needed. A
20 kHz waveform will NOT be better catpured at a sample
rate of 48 kHz than at 44.1.
> >The situation described in the paragraph above is exactly
> >the reason why you need more than two samples; at exactly
> >two the wanted and the first alias collide, and as
> >they are 180 degrees out of phase with each other, they cancel.
>
> not exactly. sure, the resultant wave would be 'nothing',
> but not because the sampling rate is out of phase with the
> frequency, but rather, perfectly in phase with the zero
> crossings. the sound pressure level at the zero crossings
> is zero (or -infinity from nominal).
> So, if the sample rate is, say 22k exactly,
> and the frequency rate is 11k exactly, depending upon when
> the sample measures the spl at that moment, you might get
> a perfect sawtooth or you might get nothing.
And you have just violated completely the tenets of
the Nyquist-SHannon sampling theorem.
Why would anyone reaosnably expect any usable results if
you violate the operating principles?
> of course, such a situation is not likely in the 'real world',
> because no natural sound source would be that perfectly
> in lock step with the sampling frequency. Nevertheless,
> 'just two' samples per wave is insufficient to faithfully
> replicate a waveform, and invites distortion.
And that's why NO ONE who knows anything says
"exactly two samples per cycle." It's ALWAYS "more
than two samples." And "more than two DOESN'T
mean three, it means ANY number more than two, like
2.000000000000000000001.
dpierce.ca...@gmail.com
> technically, any waveform requires at least two samples per wave.
No, "mathematically", to fully replicate a waveform requires
MORE than two samples of the highest component of
that wave.
> at two samples, for example, you really cannot tell
> whether the waveform is sine, square, sawtooth, or whatever
> variant the harmonics may imbue to the waveform. Even
> at four samples, the original waveform is only approximated.
> (i.e. sine can look square or sawtooth depending upon
> at what degrees of arc the sample is taken.)
More than two, that's all you neeed, for the highest
component of interest: end of discussion.
> that said, few humans can detect the difference between
> a sine, square, or sawtooth above 10k.
I would challenge you to find said few humans.
This claim is made over and over again and it's flawed
beyond utility.
The test is almost always made with a standard lab function
generator whose output waveforem are normalized to have
the same peak aimplitude regardless of the waveform., e.g.,
the sine wave is 1 V peak-to-peak and the square wave is
1 v peak-to-peak.
Set the frequency to 10 kHz, and it's actually pretty easy for
most people to tell the different between the switch set to
sine wave and the switch set to square wave, but they are
NOT hearing the difference between a 10 kHz sine and a
10 kHz square wave: They're hearing the difference between
a 10 kHz sine wave with a peak-peak amplitude of 1 volt
and a 10 kHz sine wave with a peak amplitude of about 1.28
volts. That 1.28 volt sine wave is the amplitude of the 10
kHz sine wave fundamental of your 1 volt peak square wave.
That difference, almost 2 dB, is actually quite EASY to hear.
Now, go find us these few humans that can hear the
difference between a 10 kHz 1 volt P-P sine wave and
a 10 kHz 0.786 volt P-P square wave, and now you
might have something. But since no one else has
survived the challenge, I'd not place any hard cash on
you being the first.
> But there is a difference.
Not when the signal is PROPERLY band-limited to less
than 1/2 the same rate, there isn't, other than simple in-
band amplitude differences.
> How we as humans interpret that difference
> is not easily quantifiable--some may speak in terms of
> clarity, crispness, 'air', 'musicality', or what have
> you--not very useful terms to the engineer.
It's not very quantificable because those making the
claims have never quantified it.
> As well, sampling at a fixed frequency any other
> frequency pattern will result in sampling-induced
> harmonics, non-musical, that are in fact lower
> than the fundamental of the wave. the amplitude
> of these harmonics reduces greatly with increased
> number of samples per wave, so at under, say, 5K,
> they are not noticable, when sampling at 44.1k.
Complete and utter nonsense. As long as the signal
is bandlimited to less than 1/2 the sample rate, no
such artifacts exist. Period. That means any,
repeat ANY waveform that is bandlimited to less
than 1/2 the sample rate will be captured with NO
additional artifacts.
> finally, the lower the sampling frequency, the
> increased number of artifacts created when resampling.
Not as long as it is greater than twice the abndwidth, it
won't.
> So, if one recorded some cuts at 44.1 and others at
> 48k, the movement back and forth to digitally (or analog)
> mix will have the effect of altering the higher-frequency
> waveforms in progressive generations of resampling.
Not for ANY waveform band-limited to 20 kHz, it won't.
> the bottom line is that sampling at 44.1k has
> a noticable and significant impact on all waveforms
> over 11k, and some modest impact over 5.5k.
Again, complete nonsense.
> Impacts increase with successive generations
> of resampling (any time bit rate changes, and
> in particular, compression.
> Measuring whether humans can detect such a
> change is torturously difficult.
Actually, it's not.
> prior to the 96 and 192 rates, i found that recording
> and mixing all within 44.1 resulted in less resampling
> bias and artifacts than switching between 48 and 44.1.
What on earth is "resampling bias?"
> >And if this ain't the case, why would the sampling rate be called
> >"frequency?"
>
> because the frequency of a sine waveform involves a
> continuous function that goes both positive and negative
> during one complete 'wave'. There are generally two
> zero crossings during a wave. Let's say, for a moment,
> that I sampled the audio energy at each zero
> crossing--exactly two samples per waveform, or at exactly
> 22.05 KHz.
If you do this, you have violated Nyquist/SHannon. You have
a broken, defective sampler.
Why then take something that's broken and try to make any
sense or draw any conclusion from the result: IT'S BROKEN!
> Then, upon playback, i iterpolate a straight line between
> the samples.
During playback, you NEVER "interpolate a straight line."
> What waveform would result?
Why would anyone care what waveform results from
your broken sampler? IT'S BROKEN!
Dave Platt
>>two samples per wave, not at least two samples per wave.
>
>...so, if i require "more than two per wave", then
>are you saying that the highest frequency distinguishable
>from 44.1k is something less than 15k (44.1 divided by 3)?
Any decent CD player has a frequency response which is flat to within
a small fraction of one dB, up to 20 kHz, and accurately resolves
frequencies within that bandwidth.
>>As long as you have more than two, the wave is uniquely
>>and totally described - any further samples are unneeded
>>and superfluous.
>
>only if you assume a perfect sine wave with no
>overtones.
If it has overtones (i.e. harmonics), then by definition these
harmonics have to be taken into account as part of the "highest
frequency distinguishable". For example, a non-sinusoidal 20 kHz
signal contains overtones (e.g. at 40 and 60 kHz, the second and third
harmonics) and is actually a composite signal which has 60 kHz of
bandwidth. This cannot be accurately sampled and reconstructed at a
sampling rate of 44.1 ksamples/second.
That's why signals must be low-pass-filtered before being sampled.
It's an essential part of the process.
> To achieve faithful replication of up to 20k,
>then at least 3 samples per wave at 20k would be necessary,
>true?
No, not true. You don't need an _integral_ number of samples per
cycle (e.g. 3 or 4) to accurately distinguish, and reproduce, the
signal. All that's required is that you have somewhat more than two.
With CDs, a 20 kHz audio bandwidth, and a 44.1 kHz sampling rate,
gives you a minimum of 2.2 samples per cycle (at the 20 kHz bandwidth
limit) and more than that at lower frequencies. This is sufficient
to accurately reproduce the signal.
This may seem counter-intuitive, but it actually does work (both in
practice, and in the underlying mathematics).
> I mean, 44.1 and 48k wouldn't cut it.
>only something greater than 60k. e.g. 96k.
This turns out not to be the case. 44.1 ksamples/second *does*
allow the accurate sampling, and reconstruction, of an audio signal
with 20 kHz of bandwidth.
There are two *essential* steps in this process. You *must* filter
the incoming continuous signal before you sample it, to ensure that it
actually has no more than 20 kHz of bandwidth (i.e. you must filter
out any individual signals, or harmonics/overtones which lie above 20
kHz). This is usually known as the "anti-aliasing filter" step.
Then, when you convert the samples back to continuous form, you *must*
run the samples through another bandwidth-limiting filter (again, DC
to 20 kHz in the case of CDs) to eliminate the image frequencies lying
above 20 kHz. This is usually referred to as the "reconstruction
filter".
>of course, such a situation is not likely in the 'real world',
>because no natural sound source would be that perfectly
>in lock step with the sampling frequency. Nevertheless,
>'just two' samples per wave is insufficient to faithfully
>replicate a waveform, and invites distortion.
>i would agree that 'more than two' are thus necessary.
>which, in my klunker way of putting it, was what i
>was trying to say in the first place.
You're correct. You need more than two.
You just don't need all that much *more* than two. You don't need
three. 2.2 turns out to be sufficient, if you do a proper job
implementing the anti-aliasing and reconstruction filters.
--
Dave Platt <dpl...@radagast.org> AE6EO
Friends of Jade Warrior home page: http://www.radagast.org/jade-warrior
I do _not_ wish to receive unsolicited commercial email, and I will
boycott any company which has the gall to send me such ads!
dpierce.ca...@gmail.com
> 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?)
Others have chimed in (regretably, some incorrectly), but
the short answer is: the nathematical basis behind periodic
sampling tells us that if we have a singnal whose bandwidth
is, oh, "x", to capture that sample that signal with no loss of
time-domain information or have no unwanted artifacts, we
are required to sampke that signal at MORE THAN twice "x".
For example, if you assume the bandwidth of human
hearing is 20 kHz, you must sample at more than
2*20 kHz or GREATER THAN 40 kHz to ensure that
everything within that 20 kHz bandwidth is captured
and not lost.
One of the MOST important parts of a properly implemented
sampler is the preceeding band-limiting filter. ALL operational
sampler (no exceptions) provide some means of ensureing
that NO components outside that bandwidth reach the sampler.
What it comes down to is this: IF you can assume, a priori,
that your bandwidth is LESS THAN 1/2 the sampling rate,
ALL waveforms within that bandwidth can and are uniquely
identified by all available samples. If you have, say, 2.0001
samples per cycle of, say, a sine wave, there is exactly one
and ONLY one waveform whose bandwidth is less than
1.2 the sample rate that can pass through those samples.
Adding more samples will NOT make the representation
of that waveform ANY more precise: it will simply waste data.
> 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?
Because, if you follow the discussion about the requirements
of a sampler, when you have down-sampled to 22 kHz, you
have to first filter EVERYTHING that's at or above half that,
or 11 kHz. And many people have no problem hearing the
intrusion of a <11kHz low pass filter on the right kind of
musical material.
And, to repeat, it's not doubled, it's multiplied by SLIGHTLY
more than double.
> Also, just where the
> hell did the number 44,100 emerge from? Why not 40,000?
Way back in the late 1970's, the only form of portable
recordable storage that was affordable for the kind of data
rates needed for digital audio were video tape recorders.
The samplers used the vertical modulation from white to
black to store 1's and 0's. To ease the designof the samplers
and the synchronization and to meet the bandwidth limts
of the recorders, it was decide to put an intergral number
of samples on each scan line.
For 60 Hz/525 line NTSC, you have 35 blanked lines, leaving
490 lines per frame, 245 line for field. Storing 3 samples per
line, the resulting sample rate becomes:
60 field/second * 245 lines/field * 3 samples/line =
44.1 kHz samples per second. Similar calculations yield
sample rates of 48 kS/s and 50 Hz/625 PAL video can also
accomodate these rates in similar fashions.
One very interesting side benefit is not only could you
store wide-band audio this way, you could also transmit
the resulting digitized audio over normal broadcast TV
channels with no loss. During the '80s' a number of stations
did just that: if you had a compatible D/A converter, you
could listen to full bandwidth digital audio at home from your
TV set.
> 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?"
Becasue "frequency, in the technical parlance, means quite
precisely "per unit time" Whether it's cycles per second or
sample per second or high tides per day, they all describe
the frequency, or how often, at which some semi-periodic
event happens.
Don Pearce
> Don Pearce wrote:
>
>> steve wrote:
>>> because the frequency of a sine waveform involves a
>>> continuous function that goes both positive and negative
>>> during one complete 'wave'. There are generally two
>>> zero crossings during a wave. Let's say, for a moment,
>>> that I sampled the audio energy at each zero
>>> crossing--exactly two samples per waveform, or at exactly
>>> 22.05 KHz. Then, upon playback, i iterpolate a
>>> straight line between the samples. What waveform
>>> would result?
>>>
>>> -steve
>
>> Utter cock.
>
> ok.
>
>> And of course sampling theory states that you need MORE THAN
>> two samples per wave, not at least two samples per wave.
>
> ...so, if i require "more than two per wave", then
> are you saying that the highest frequency distinguishable
> from 44.1k is something less than 15k (44.1 divided by 3)?
>
>> As long as you have more than two, the wave is uniquely
>> and totally described - any further samples are unneeded
>> and superfluous.
>
> only if you assume a perfect sine wave with no
> overtones. To achieve faithful replication of up to 20k,
> then at least 3 samples per wave at 20k would be necessary,
> true? I mean, 44.1 and 48k wouldn't cut it.
> only something greater than 60k. e.g. 96k.
>
This is where you need to understand a bit of theory, and maybe it isn't
all that intuitive. If you have a signal that is limited (by filtering)
to a bandwidth less than 22.05, then those two-and-a-bit samples per
wave describe it perfectly and unambiguously. There is only one solution
possible for the DAC to work on, and that is the identical wave that was
present at the input.
>> The situation described in the paragraph above is exactly
>> the reason why you need more than two samples; at exactly
>> two the wanted and the first alias collide, and as
>> they are 180 degrees out of phase with each other, they cancel.
>
> not exactly. sure, the resultant wave would be 'nothing',
> but not because the sampling rate is out of phase with the
> frequency, but rather, perfectly in phase with the zero
> crossings. the sound pressure level at the zero crossings
> is zero (or -infinity from nominal).
> So, if the sample rate is, say 22k exactly,
> and the frequency rate is 11k exactly, depending upon when
> the sample measures the spl at that moment, you might get
> a perfect sawtooth or you might get nothing.
>
Could be zero crossings, or any other point up and down the wave -
depends on where the sampling happens during the cycle.
> of course, such a situation is not likely in the 'real world',
> because no natural sound source would be that perfectly
> in lock step with the sampling frequency. Nevertheless,
> 'just two' samples per wave is insufficient to faithfully
> replicate a waveform, and invites distortion.
>
> i would agree that 'more than two' are thus necessary.
> which, in my klunker way of putting it, was what i
> was trying to say in the first place.
>
No, you were saying that much more than 2 were needed. That simply isn't so.
d
us...@domain.invalid
>
> Now, go find us these few humans that can hear the
> difference between a 10 kHz 1 volt P-P sine wave and
> a 10 kHz 0.786 volt P-P square wave, and now you
> might have something. But since no one else has
> survived the challenge, I'd not place any hard cash on
> you being the first.
> \
That's a TOUGH test, since the 2nd harmonic (at 20 kHz)
is missing from a square wave. It's also missing from
a symmetric triangle wave. The lowest harmonic present is at 30 kHz.
A better test would be a 10 kHz sine wave with an
added second harmonic (i.e. 20 kHz.)
Nevertheless, I HAVE heard the difference between a 10 kHz
since wave and a 10 kHz square wave with the same amount of
the 10 kHz component, on a speaker. AND I actually know WHY
I heard it. Using a 100 kHz bandwidth HP instrumentation
mike and a computer with a 200 kHz sampling rate National
Instruments card, with a 80 kHz multiple low
pass filter, it is quite clear the the speaker I used
was generating not-harmonicly-related buzzing due to
nonlinearities, as well as having the added harmonics
change the level of the produced 10 kHz. This was a
very old speaker, the tweeter from an AR3. I could also
hear the difference between a 13 kHz sine wave and one with
added 2nd harmonic, even though I could not hear the
26 kHz (which was there), for the same reason. The speaker
was rather badly down in response above 20 kHz, but not to zero.
And that's not even worrying about ordinary IM distortion
(say, between a 20 kHz sine wave at its alias at 24.1 kHz,
which is of course at 4.1 kHz) in speakers.
Speakers matter.
Doug McDonald
Randy Yates
> 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?
What you're missing is that the bandwidth of a digital system is
HALF the sample rate. So sampling at 44.1 kHz passes (potentially)
a signal with frequencies up to 22.05 kHz.
> Also, just where the hell did the number 44,100 emerge from? Why not
> 40,000?
See 44,100 and 44,056 here:
http://en.wikipedia.org/wiki/Sampling_rate
> Nowadays, DVD-audio songs are recorded at 96/192 KHz, is there a
> point?
Yes - to make Sony and other media moguls more money (by requiring
people to replace their collections). Other than that, no.
> And if this ain't the case, why would the sampling rate be called
> "frequency?"
Simple laziness - humans get lazy with terms.
--
% Randy Yates % "Ticket to the moon, flight leaves here today
%% Fuquay-Varina, NC % from Satellite 2"
%%% 919-577-9882 % 'Ticket To The Moon'
%%%% <ya...@ieee.org> % *Time*, Electric Light Orchestra
http://www.digitalsignallabs.com
Chronic Philharmonic
"Randy Yates" <ya...@ieee.org> wrote in message
news:m3lk0es...@ieee.org...
> Industrial One <industr...@hotmail.com> writes:
[snip]
>> And if this ain't the case, why would the sampling rate be called
>> "frequency?"
>
> Simple laziness - humans get lazy with terms.
I do not think the term frequency is being used improperly. A sample rate
has a frequency:
Q: "How frequently are you sampling?"
A: "44.1 thousand times a second -- at a frequency of 44.1KHz"
glen...@gmail.com
Here is a good article for understanding sampling theory:
http://www.wescottdesign.com/articles/Sampling/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.
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. So
depending on what analog filters cost, etc. etc. there might be some
sense in going with 96khz systems. 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).
2- Anyways this is just speculating. The real way to figure it out is
to do a test. Unfortunately I haven't done so myself. But according
to one audio engineer, there is an audible difference. So maybe there
is merit to 96khz systems. Do read the sampling article as it
provides a better understanding of what goes on.
http://www.prorec.com/Articles/tabid/109/EntryId/158/Default.aspx
QUOTE:
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.
Pasi Ojala
> 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.
Take a square wave for example. You can create that as a digital
signal "easily", by taking a base-frequency sine and all of the
odd multiplies with the proper gains. Continue into infinity and
you have a nice square wave. Stop at the band edge and you get
"ringing". But it really isn't an artifact, you have all the
sines intact that you added.
> On reproducing that signal, there is that set of tradeoffs a second
> time. So you can lose frequency response there again.
You can lose bandwidth, but there is enough information to
perfectly generate the signal that was properly low-pass
filtered at the sampling end.
> depending on what analog filters cost, etc. etc. there might be some
> sense in going with 96khz systems. It definitely does make sense to
> sample at 96khz at acquisition... the oversampling is beneficial
Analog filters have not been used for years. You digitally
upsample and filter to get a DAC frequency of for example 6MHz,
then use a first-order lowpass to filter above for example 100kHz.
There is no need to use 96kHz samplerate.
> To hell with theory, my EARS tell me there is a difference.
Everything else being equal you can't hear frequencies above
22kHz. So obviously everything else has not been equal.
-Pasi
--
"I know, I know what it's like to lose someone, only to find her and
then lose her a second time. I wouldn't wish that on anyone, not even
you, as much as I might want to."
-- Sheridan to Bester in Babylon 5:"Rising Star"
dpierce.ca...@gmail.com
> dpierce.cartchunk....@gmail.com wrote:
>
> > Now, go find us these few humans that can hear the
> > difference between a 10 kHz 1 volt P-P sine wave and
> > a 10 kHz 0.786 volt P-P square wave, and now you
> > might have something. But since no one else has
> > survived the challenge, I'd not place any hard cash on
> > you being the first.
> > \
>
> That's a TOUGH test, since the 2nd harmonic (at 20 kHz)
> is missing from a square wave. It's also missing from
> a symmetric triangle wave. The lowest harmonic present
> is at 30 kHz.
But it is the test often cited as "proving" that one can hear
the difference. It's not my choice of a test, to be sure, and
becasue of the way it's normally suggested it be conducted,
it's a trivially easy test to shoot down because of its flaws.
Yet we see it being brought up over and over again.
> A better test would be a 10 kHz sine wave with an
> added second harmonic (i.e. 20 kHz.)
>
> Nevertheless, I HAVE heard the difference between a 10 kHz
> since wave and a 10 kHz square wave with the same amount of
> the 10 kHz component, on a speaker. AND I actually know WHY
> I heard it. Using a 100 kHz bandwidth HP instrumentation
> mike and a computer with a 200 kHz sampling rate National
> Instruments card, with a 80 kHz multiple low
> pass filter, it is quite clear the the speaker I used
> was generating not-harmonicly-related buzzing due to
> nonlinearities, as well as having the added harmonics
> change the level of the produced 10 kHz.
But, of course, you're not hearing the difference between
a 10 kHz sine wave and a 10 kHz square wave, you're
hearing the difference between a 10 kHz sine wave and
some other wave, BOTH of which have components
within the audible bandwidth, and as such do nothing
to support the case that the information above 20 kHz
has an audble effect.
> And that's not even worrying about ordinary IM distortion
> (say, between a 20 kHz sine wave at its alias at 24.1 kHz,
> which is of course at 4.1 kHz) in speakers.
The alias would only occur anyway if the system was badly
implemented. Further, on musical material it's not likely
that you will EVER find energy at high enough frequencies
such that these sorts of artifacts will have a sufficient level
to overcome the masking effects of that material that's
already there to begin with.
Claims of these kinds suffer from all sorts of serious
and funcdamental problems. It suggests that many of
the claimants and proponents of wide-band audio touting
them have very likely NEVER conducted the experiments
and very likely never even heard a situation where they
MIGHT occur, because the basis of the claims themselves
are so dubious and poorly thought out.
dpierce.ca...@gmail.com
> 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.
A fundamental error here.
First, sampling does not cause the "ringing," it's the
truncation of the bandwidth. You can have all sorts of
ringing in a continuous time analog system. Look at the
output, for example, of an old-style analog anti-aliasing
filter that preceeded early generation A/D converters.
Take a 1 kHz square wave. Band limit it with a 20 kHz
low-bass filter.
Now, sum the following series:
F(t) = sum (sin(x*t)/x) where x = 2 pi * 1, 3, 5, ... 19
and see what you get. Is the ringing in the first case real
and in the second case simply a result of truncation of
a mathematical series?
> On reproducing that signal, there is that set of tradeoffs a second
> time. So you can lose frequency response there again.
Why? How? I have inexpensive A/D and D/A chains here
that have frequency response from 2 Hz to 20 kHz with less
than +-.2 dB total error across the band and with a phase
response 20-20 kHz with 5 degrees or 0.
What "lost frequency response" are you talking about?
> 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.
I don't sample it again and again when I reproduce it.
> So depending on what analog filters cost, etc. etc. there
> might be some sense in going with 96khz systems.
The analog filter in the BEST 44.1 kHz digital system I have
here costs on the order of a buck or two and has almost
NO imact whatsoever on the frequency response within
the 20 kHz audio bandwidth.
> 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).
That's why NO ONE uses analog filters to do this job.
THat's why no one with any competence has used analog
anti-aliasing and anti-imaging filters for two decades.
And what you're talkig about is NOT "oversampling."
In fact, for the last two decades, MOSTY A/D and D/A
systems HAVE used oversampling techniques to
elininate the issues surrounding analog filters. And the
run not at 96 kHz or 192, but are 44.1 or 48 kHz oversampled
systems.
> 2- Anyways this is just speculating. The real way to figure it out is
> to do a test. Unfortunately I haven't done so myself. But according
> to one audio engineer, there is an audible difference.
And, according to many independent researchers, there is not.
Earl Kiosterud
"cr88192" <cr8...@NOSPAM.hotmail.com> wrote in message
news:a2339$4870bc16$ca83b482$13...@saipan.com...
>
> "Industrial One" <industr...@hotmail.com> wrote in message
> news:591b3f55-e2e1-408e...@z72g2000hsb.googlegroups.com...
> for something unrelated it is a thought that, for "audio as data" uses, one could stuff a
> little bit higher-range data into the 16-bit samples, by representing a slightly bigger
> range (say, 24 bits), as log-scaled values within the 16 bit range. of course, this would
> likely reduce quality a little, and add a little noise, if played back as 16 bit audio
> (mostly, really quiet stuff that would normally be quantized away is left).
>
Along a similar line, I've wondered why we didn't non-linearize the the digital audio values
for 16-bit audio. You get less noise at low signal levels,, though more at higher levels
where it would be masked. Back in the 70's, I was experimenting with 8-bit ADC and DAC
chips, which were a bit noisy, as you would expect for 8 bits. The noise was constant with
regard to signal level (it sounded a lot like a damaged speaker). I rigged up a
non-linearizing circuit before the ADC (it looked somewhat like very soft clipping), and a
complementary one after the DAC. The noise was substantially less noticeable. The quiet
parts of the audio had less noise, and the louder parts, even though the noise was actually
worse, didn't really sound so, because of masking. The function couldn't quite be called
logarithmic, as the values went through zero, but were close for most values.
Floating-point audio achieves something similar.
If nonlinearizing the transfer function was considered when CD audio was being standardized,
I suspect that one reason it was rejected was that it would open arguments about the
accuracy of the system. The complementary function in a CD player would have to exactly
match that of the CD, or there would be amplitude distortion. I think it'd have to have
been done in analog -- doing it in the digital domain would ensure compatibility, but wasn't
practical back then -- it would have required DSPs and converters with greater than 16 bits,
impractical for the time for consumer stuff.
--
Earl
Don Pearce
This is done already for telephone lines - and very successfully. There
are two system - A-Law and Mu-Law; the first is international and the
second local to the USA. If there were such a system applied to 16-bit
systems, we could be seeing better than 24 bit performance. But first
agreeing and then constructing the necessary variable slopes was way
beyond the technology of the day.
It could still be done very simply during a bit-reduction process from
24 to 16 bits - just a mathematical operation. Then at the player end
those 16 A-law bits could be reconstructed back to 24 very quiet bits.
Of course all this would save is some real estate on the recording
medium. Interesting project, though.
d
us...@domain.invalid
>> If nonlinearizing the transfer function was considered when CD audio
>> was being standardized, I suspect that one reason it was rejected was
>> that it would open arguments about the accuracy of the system. The
>> complementary function in a CD player would have to exactly match that
>> of the CD, or there would be amplitude distortion. I think it'd have
>> to have been done in analog -- doing it in the digital domain would
>> ensure compatibility, but wasn't practical back then -- it would have
>> required DSPs and converters with greater than 16 bits, impractical
>> for the time for consumer stuff.
>
> This is done already for telephone lines - and very successfully. There
> are two system - A-Law and Mu-Law; the first is international and the
> second local to the USA. If there were such a system applied to 16-bit
> systems, we could be seeing better than 24 bit performance. But first
> agreeing and then constructing the necessary variable slopes was way
> beyond the technology of the day.
>
That would not work in a single-band system. Consider a piece of music
with a high loudness bass drum and a very soft flute. The high absolute value
part of the wave of the drum would be in a part of teh amplitude regime
where the sampling points (in amplitude) were far apart, and the flute
signal might be missed entirely when the bass amplitude was low.
That's why MP3 is an agressively multiband system. It **IS** such
a system, which works well.
Doug McDonald
Don Pearce
Huh?
d
Industrial One
> 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?
> 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
Pffft... I'm 18 and I can hear 17 KHz maximum, assuming 20 KHz for
anyone is an exaggeration, and whoever claims to tell the difference
between 44/96 is some autistic motherfucker that probably pisses
himself in class 'cuz he keeps hearing the "whistling" from the rat
repellant. Also, I've subtracted a 44 track by a downsampled 32
version to hear what EXACTLY is stripped and all I heard was extremely
faint clicks, jingles and dings (from the beat of the treble
percussion instruments) and had to amplify the waveform up 30 dB to
hear it clearly (real headache inducer, yo.) So... what I'm really
missing from a 96 KHz track are some faint jingles TWICE as high-
pitched and inaudible from the upper freqs of a 44 KHz track?
WOW........ is THIS really what those assramming audiophiles bitch
about?
> 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
Proof drugs are awesome.
> is better stuff, and has been more meticulously recorded, hence the good sound of many of
> them. It ain't the extravagant bit depth and sampling rate. There are some damned
> good-sounding CDs too. Even if you had a regular CD version and a DVD-audio version, and
> the DVD-audio version sounded better, would you actually believe that the improvement was
> because of the bit depth and sample rate? Couldn't be anything else, could it? How are
> they going to sell DVD-audio discs if they let the CDs sound the same?
QFT.
> Hope this helps.
> --
> Earl
Was informative, thanks.
On Jul 5, 10:31 pm, "Richard Crowley" <rcrow...@xp7rt.net> wrote:
> The "Red Book" convention for making audio CDs was developed
> back in the early 1980s and established 44.1KHz as the sampling
> rate. In order to maintain forwards and backwards compatibility,
> all CDs must use that sample rate.
>
> Any periodic occurance can be referred to as "freqency". Whether
> it is something that happens every femtosecond (like light) or every
> 1000 years (like the century).
>
> The Nyquist-Shannon sampling therom tells us that you must
> sample at *twice* the desired highest frequency to adequately
> reproduce the original waveform. It is said that 22KHz was
> selected as the top end (x2 = 44KHz) because of the state of
> the art in filters back in those days.
I see.
> > Obviously, one can notice the difference if the song was
> > downsampled to 22,
>
> You notice it because reducing the sampling rate to 22KHz
> actually reduces the top end to 11KHz which many people
> can detect.
I don't follow. Why 11?
On Jul 6, 2:06 am, "Chronic Philharmonic" <karl.uppi...@verizon.net>
wrote:
> It was because they used video recorders for mastering prototype and first
> generation CDs, and it was the nearest available frequency that was greater
> than 40KHz needed to meet the sampling Nyquist requirement of at least two
> samples for the highest frequency to be recorded (20KHz). See also:http://www.cs.columbia.edu/~hgs/audio/44.1.html
Damn, I didn't know that.
> > Nowadays, DVD-audio songs are recorded at 96/192 KHz, is there a
> > point?
>
> Marketing. There is no defensible mathematical requirement for it.
's what I thought. I wouldn't be surprised if them retard scene
rippers start releasing audio in 192 KHz with a bitrate of probably
1024+ and advertise "DVD QUALITY AUDIO!"
I'll read the other posts later.
Willem
) Do I understand correct: hz is one sine loop per second,
Hz is one <whatever> per second.
Your heart beats at roughly 80 Hz.
The moon revolves around the earth at roughly 0.38 uHz (Microhertz)
) 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?
Play your sample at half speed and listen to what happens when
you reach 11.025 Khz.
)> > Obviously, one can notice the difference if the song was
)> > downsampled to 22,
)>
)> You notice it because reducing the sampling rate to 22KHz
)> actually reduces the top end to 11KHz which many people
)> can detect.
)
) I don't follow. Why 11?
Because 11 is half of 22. See above.
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
Earl Kiosterud
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.
A sine has only one frequency component in the spectrum, and a single frequency component in
the spectrum is a sine.
There's your sine.
--
Earl
dpierce.ca...@gmail.com
> Do I understand correct: hz is one sine loop per second,
No, "Hz" simply is the abbreviation for "Hertz" which
is the SI unit for reciprocal second. It has nothing to do
per se with sine waves, square waves, are waves of
any kind,. Rather it is simply the rate at which whatever
semi-periodic phenomenon occurs. It could be the
rotational period of a neutron star, like 792 Hz.
> 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?
Uhm, I think, if I am able to parse what you are saying, yes,
though your grammar is a bit tortured.
But a couple of points:
What you "see" in an audio application often does NOT,
depending upon the application, reflect what the resulting
weaveform actually looks like. The EASIEST graphic display
of a series of sample valies is to simply connect the dots,
but that is NOT what D/A converters do. If you had a
series of alternating +- and -1 values, near 1/2 the sample
rate, and if simply "drew" them by connecting the dots,
what you would see could be what looked like a triangle
wave carrier modulated by a sine wave. FOr example,
at 22 kHz, you'd see the modulating wave as having a
frequency of 50 Hz.
But that waveform, AS DRAWN, has components that
extend to infinity. JUstv as there is a component at
22.05 kHz - 50 Hz = 22 khz, there is also one at 22.05+50 Hz
or 23 kHz. If you had only three such components, then
the waveform would be a 22 kHz SINE wave modulated by
50 Hz sine. But the triangle carrier has compoenents
way out as well, so you'll see images at 44.1 +- 50 Hz, and
88.2 kHz +- 50 Hz and so on.
ALL these are IMAGES of the original 22 kHz sine wave and,
to be reconstructed properly, requires the additon of the
necessary anti-IMAGING filter (also known as the reconstruction
filter). Now, get rid of all the components at an above 22.05
kHz, and all you're left with is your original 22 kHz sine, just
as it was recorded.
Now, where OVERsampling techniques come in is HOW
you implement this filter. YOu can, as some assumed incorrectly,
try to implement it in an analog filter, and that get's real tough.
You can also try to implement it as a digital filter at 44.1 kHz,
but it's effectively impossible to implement a 22.05 kHz low
pass filter at a sample rate of 44.1 kHz.
So what we do is we OVERsample the 44.1 kHz stream that
we already have: inbetween every real sample, lets insert, oh
63 samples whose value is, let's say, 0. Or anything slse, for
that matter.
What we've done is that we've taken the original images
spaced 22 kHz apart and now spaced them instead
22 kHz * 64 or 1411 Khz apart.
Now you can build a VERY GOOD digital filter with very
good out-of-band rejection and very good in-band response,
and then all you have to do in your final analog stage is
make sure 20 kHz is getting through fine, but 1411 Khz is
properly attenuated.
That's a LOT easier than building a filter which lets
20 Khz through but attentuates 22.05 kHz completely.
Earl Kiosterud
"Don Pearce" <nos...@nospam.com> wrote in message
news:GaOdnfePq5f1iO_V...@posted.plusnet...
stuff.
>
> This is done already for telephone lines - and very successfully. There are two system -
> A-Law and Mu-Law; the first is international and the second local to the USA. If there
> were such a system applied to 16-bit systems, we could be seeing better than 24 bit
> performance. But first agreeing and then constructing the necessary variable slopes was
> way beyond the technology of the day.
>
> It could still be done very simply during a bit-reduction process from 24 to 16 bits -
> just a mathematical operation. Then at the player end those 16 A-law bits could be
> reconstructed back to 24 very quiet bits. Of course all this would save is some real
> estate on the recording medium. Interesting project, though.
>
> d
I think I read somewhere that AT&T was using digital for long-distance as early as 1970,
long before most of use had even heard of digital audio. And it was 8-bit, 8 Khz sample
rate, I think.
To do the bit-reduction method, it seems that something like a 24-bit DAC would be needed in
the player.
In my project, I didn't try changing the nonlinearization much, and then found that the low
order bit of my 8-bit chips didn't appear to be working correctly anyway. Mine was a
continuously variable slope, using transistors in op-amp feedback loops. It was only a
for-fun project anyway, so I didn't pursue it further to see how much I could reduce the
noise -- I didn't have an application for it at the time. But it did prove the
nonlinearization workable, even if it didn't achieve AT&T's quality.
--
Earl
Peter Schepers
Willem <wil...@stack.nl> wrote:
>Industrial One wrote:
>) Do I understand correct: hz is one sine loop per second,
>
>Hz is one <whatever> per second.
>Your heart beats at roughly 80 Hz.
I think you mean the heart beats 1.333 Hz.
PS.
Willem
) In article <slrng74cnd...@snail.stack.nl>,
) Willem <wil...@stack.nl> wrote:
)>Industrial One wrote:
)>) Do I understand correct: hz is one sine loop per second,
)>
)>Hz is one <whatever> per second.
)>Your heart beats at roughly 80 Hz.
)
) I think you mean the heart beats 1.333 Hz.
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.
Don Pearce
Better do some exercise then. Mine is more like 0.85Hz.
It has to be said, though, that in virtually every calculation that
involves frequency, the number you actually want is radians per second -
2pi * normal frequency. It would make life much easier if the world
adopted that measure.
d
Peter Schepers
Don Pearce <nos...@nospam.com> wrote:
>Peter Schepers wrote:
>> In article <slrng74cnd...@snail.stack.nl>,
>> Willem <wil...@stack.nl> wrote:
>>> Industrial One wrote:
>>> ) Do I understand correct: hz is one sine loop per second,
>>>
>>> Hz is one <whatever> per second.
>>> Your heart beats at roughly 80 Hz.
>>
>> I think you mean the heart beats 1.333 Hz.
>>
>
I was using his 80 beats/min, not mine.
PS.
Peter Schepers
Willem <wil...@stack.nl> wrote:
>Peter Schepers wrote:
>) In article <slrng74cnd...@snail.stack.nl>,
>) Willem <wil...@stack.nl> wrote:
>)>Industrial One wrote:
>)>) Do I understand correct: hz is one sine loop per second,
>)>
>)>Hz is one <whatever> per second.
>)>Your heart beats at roughly 80 Hz.
>)
>) I think you mean the heart beats 1.333 Hz.
>
>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...
Well, I originally calc'd .41uHz instead of your number.
PS.
Steven Sullivan
> 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.
The 'hi rez' format where you are more likely to get 'full-range' dynamics is SACD, due to
restrictions built in to the Scarlet Book spec. But even there it's now possible to squeeze
the range, from what I hear.
In any case, there's no reason why these full-range recordings couldn't be presented on
good old 16-bit CD. The perversion of one of the original benefits of Redbook (extended
dynamic range) into the 'new' benefit (heretofore unheard-of amounts of compression)
is just sad.
--
-S
Poe's Law: Without a winking smiley or other blatant display of humorous
intent, it is impossible to create a parody of a religious Fundamentalist that
SOMEONE won't mistake for the real thing.
Steven Sullivan
> 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 same problem with electrostatic
> speakers with step up transformers . So no matter how much better
> SACD or DVDA can be, played over such loudspeakers all the
> advantages are down the drain.
> Don't know if this story is true but I very much like to attend such
> a listening test an judge for myself.
> Did anyone here did attend such a test and on what kind of speakers
> was it played?
No doubt the test is being 'swift-boated' by anxious 'audiophiles', but here are the facts
(clipped from
http://www.bostonaudiosociety.org/explanation.htm )
The Principal System
The playback equipment in this system consisted of an Adcom GTP-450 preamp and a Carver M1.5t
power amplifier. Speaker cables were 8 feet of generic 12-gauge stranded wire; the line-level
connecting cables were garden-variety. Three different players were used: a Pioneer DV-563A
universal player, a Sony XA777ES SACD model, and a Yamaha DVD-S1500. The loudspeakers were a
pair of Snell C5s. The CD-standard A/D/A loop was an HHB CDR-850 professional CD recorder
.
.
System 2
We also conducted a series of tests at a local CD/DVD mastering facility. I do not currently
have a detailed equipment list for this venue, but the speakers were very large and capable
high-end monitors, approximately 7 feet tall, and the power amps were sufficient to drive the
speakers to very high levels without audible distortion. Some of the source material for these
trials was a classical production which was then in process at this establishment. Like all
the others, these trials, which were done under a promise of anonymity made to those involved,
produced no significant correlations on music at normal levels.
.
.
System 3
Another series of trials took place at a facility at the University of Massachusetts - Lowell
campus, using students in their recording program as subjects. Their large monitoring room is
custom-designed and has very good acoustics, with a system to match. The system has a center
channel and surrounds, but as in the other trials we restricted ourselves to the two-channel
versions of our sources, so only the left and right were working. The equipment list for the
two channels is:
Klark Teknik DN-410 custom-modified 2-channel parametric equalizer
Stage Accompany PPA-1200 Dig Control class AB amplifier w/ crossover card
Stage Accompany ES-20 Class G amplifier w/crossover card
SLS S1266 3-way monitor (two 12" dynamic drivers, two 6" dynamic drivers, one 6" ribbon
tweeter)
Bag End ELF-1 8-Hz 2-channel low-frequency integrator
Bag End D18E-I dual-18" ELF subwoofer system
This is another professional monitoring system, installed in a large custom-built listening
room with auditorium-type seating. It was capable of very high levels with no audible
distortion as well as imaging of a quality not usually found in large spaces of this kind. We
were interested to find that our informal high-frequency-hearing tests, which we administered
to most of our subjects, indicated that these students had taken unusually good care of their
hearing. Most of them had an upper limit in our test of 16 to 18 kHz
.
.
.
System 4
Another set of trials was performed during the evening at another suburban location . a
custom-built listening room with good acoustics (with the help of an assortment of
professional absorbers and diffusers), very low background noise, and equipment that we
trusted would pass muster with most audiophiles:
Denon 2900 Universal Player with full PartsConnexion mods
Conrad-Johnson 17 LS line stage preamp
Sim Audio Moon 7 monoblock power amplifiers
Quad ESL 989 electrostatic speakers
Muse Model 18 subwoofer, 24 dB/octave crossover @ 50 Hz
Nordost SPM interconnects and speaker cable
//
Industrial One
> 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.
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 shit 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
news:72cb2187-c9c3-4fed...@t54g2000hsg.googlegroups.com
> To take a contrarians position here:
>
> Here is a good article for understanding sampling theory:
> http://www.wescottdesign.com/articles/Sampling/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
> To take a contrarian position here:
>
> Here is a good article for understanding sampling theory:
> http://www.wescottdesign.com/articles/Sampling/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
> In rec.audio.tech Earl Kiosterud <som...@nowhere.com> 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
<591b3f55-e2e1-408e...@z72g2000hsb.googlegroups.com>,
Industrial One <industr...@hotmail.com> 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
news:XOKdnf5nMaKGCe_V...@giganews.com...
> 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
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
> 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
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
>
> 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 shit 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
> 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
> > 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
> "Edmund" <nom...@hotmail.com> wrote in message
> news:48708734$0$17237$bf49...@news.tele2.nl
>
>> 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
> On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:
>
>> "Edmund" <nom...@hotmail.com> wrote in message
>> news:48708734$0$17237$bf49...@news.tele2.nl
>>> 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
news:slrng73qof...@mustatilhi.cs.tut.fi
> On 2008-07-07, glen...@gmail.com <glen...@gmail.com>
> 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
> "Edmund" <nom...@hotmail.com> wrote in message
> news:48733c04$0$451$bf49...@news.tele2.nl
>> On Sun, 06 Jul 2008 07:04:39 -0400, Arny Krueger wrote:
>>
>>> "Edmund" <nom...@hotmail.com> wrote in message
>>> news:48708734$0$17237$bf49...@news.tele2.nl
>
>>>> 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
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
>
> 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
> 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
> 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
<ckoz...@snet.net> 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
> 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" <ge...@nospam-paf.co.nz> wrote in message
news:jdOdnSNoyt34c-_V...@giganews.com...
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
) 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
> > 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" <rcro...@xp7rt.net> wrote in message
news:AredncgU06sW3-nV...@posted.pcez...
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
news:48746dc9$0$30466$afc3...@news.optusnet.com.au...
> 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
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
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
dpierce.ca...@gmail.com
>>>> The audio CD sample rate is set at 44,000Hz(44kHz) for a simple
>>>
>>> Where/who did you get that ridiculous idea from ?
> 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
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
<ckoz...@snet.net> 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?
Richard Crowley
> 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_Book_%28audio_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" <pfr...@covad.net> wrote in message
news:f4Odnbbjz--HL-nV...@supernews.com...
> > 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.
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
) 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.
ChrisCoaster
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
dpierce.ca...@gmail.com
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
>
> - 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
> 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
>
> - 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
dpierce.ca...@gmail.com
>>> 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
>
> easy on the math!
Any explanation that's CORRECT and ACCURATE
involves math beyond addition and subtraction:
assuredly multiplication and division for one.
However, perhaps an explanation by analogy might
help.
Imagine one of them ol' western movies, where the
indians on horses are chasing the settlers in their
covered wagons. Did you ever sometimes notice
that the wheels seem to be turning BACKWARDS?
Even backwards slowly?
That's due to a phenomenon known as "aliasing."
It's a direct result of having the wheel spin too fast
for the movie camera. What's actually happening
is quite simple. Pretend our movie camera is running
at 25 frames per second. And let's also preten that
ONE of the wagon wheel spokes is painted white and
all the rest are dark wood colored.
If the wagon wheel rotates slowly, sy, one revolution
per second, in one second, we will have captrued 25
images of the wheel, each one with the white spoke
rotated about 14 dgrees further than the previous frame.
When I play this movie back, and measure the wheel,
I will measure its rotation at 1 per second
If I speed it up to 2, 3, 4, 5 revolutions per second,
it's ALWAYS the case that on playback, the image
of the wheel is spinning at the right rate in the right
direction.
But now let's see what happens if it's spinning
EXACTLY 12.5 revolutions per second (gee,
that's HALF the frame rate. At that point, the wheel
spins just fast enough to make it halfway around.
So in the first frame it's pointing up. in the second
it;'s pointing down, the third up, and so on.
On playback, how fast and in what direct is the wheel
turing? Well, it's impossible to tell, because all you
see is two white spokes pointing up and down, not
moving at all.
And that happend at 1/2 the frame rate of the camera.
But slow it down JUST a little, say 10 revolutions per
second. in the first frame, say, it's pointing up. By
the second frame, it's moved 144 degrees, so it's
NOT quite poiting down. Third frame, it's moved a
total of 288 degrees and has NOT quite returned to
it's start.
Play this back, and you will see the wheel turning
at the right speed and in the right direction.
Now speed it up a little, say to 15 revolutions per
second. Now the wheel has moved just past the
bottom, 216 degrees, then 72 dgrees beyond the
start point, and so on.
And what you'll find is that NOW the wheel, instead
of spinning dorwards at 15 RPM, looks like it's
spinning BACKWARDS at 10 RPM.
As an extreme case, consider the wheel spinning
forwards at 24 RPM: because your movie camera
is not sampling fast enough, the wheel will actually
look like it's moving BACKWARDS at a measily
1 revolution per second.
What's the FAsTEST the wheel can turn before the
playback does NOT accurately depict it? It HAS to
be spinning LESS THAN 1/1 the frame rate of the
camera.
Conversely, how fast does you camera have to
run if it want's to accurately capture a wheel spinning,
say 20 revolutions per second? Well, it HAS to be
MORE THAN twice that, or FASTER than 60 frames
per second.
Now, the interesting thins about aliasing like this
is that ALL possible aliases get folded down. With
your frame rate of 25 per second you;'' get the same
picture back again with the wheel spinning at 1 rpm,
26 rpm, 51 rpm, 76 rpm, 101 rpm and so on on,
and at 24 RPM, 49 RPM, 74 RPM 99 RPM, the wheel
will look like it's spinning BACKWARDS at 1 RPM.
Notice also that aliases are occuring at MULTIPLES
of the original sample (or frame) rate +- 1 RPS, e.g.,
25*2 = 50+-1, 25*3+-1, and so on, up to infinity.
Do the experiment yourself in your head: What's the
fastest the wheel can spin without seeming to stop,
spin backwards, or spin at a rate different on playback?
The answer is ALWAYS, less than 1.2 the sample rate.
ALWAYS (assuming it's the baseband you want).
And, thank you for your concern, but your ability to
"feel my heart racing through your high-speed
connection" needs some serious reevaluation.
Your ability to sense my stree, kind sir, is quite
defective.
It has been said, however, that I'm not first in line
for the Nobel prize in suffering fools gladly. Especially
when this is easily a topic that's been discussed
HUNDREDS of times before.
geoff
>>> 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
Simple. To accurately recontruct a waveform, you need two samples of the
highest frequency specificied. Look up Nyquist Theory on you beloved Wiki -
even Wiki has that right !
Nothing to do with 20KHz worth or 'up' and 20KHz worth of 'down'.
geoff
Arny Krueger
news:2445952f-415a-45d6...@f63g2000hsf.googlegroups.com
> 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.
Sort of. So what?
To me the roll-off between 20 KHz and 22.05 KHz is not *instantaneous*.
There is a discernable, measureable slope.
For example, on a Behringer UCA 202, response is less than 1 dB down at 20
KHz, but its only about 4 dB down at 21 kHz, and 8 dB down at 21.8 kHz.
Measuring response between 21.8 and 22.05 kHz gets a little hairy, but if
you want to do the work, the slope is there.
High frequency response is a little like clipping. The debate over soft
roll-offs and sharp roll-offs is like the debate over soft clipping versus
sharp clipping.
Would you rather have a power amp that rises smoothly to 10% THD at 50 watts
and clips at 100 watts, or would you prefer a power amp that is almost
perfectly clean up to 99 watts, and clips at 100 watts? Well, as long as
you stay below 100 watts, you'd probably want the amp that is clean up to 99
watts. With a reasonable application, avoiding going over even 50 watts is
completely doable.
So, would you rather have a recorder that drops off smoothly to 1 dB down at
10 kHz, and is 3 dB down at 15 kHz, or would you rather have a recorder that
is nearly perfectly flat to 20 kHz, and then the response drops off pretty
sharply. I can tell you that if you did an ABX test, the recorder that is 3
dB down at 15 kHz will be pretty easy to pick out, and the one that is
almost perfectly flat to 20 Hz will be very, very tough.
As far as the So what part goes, here's what I mean. Does the shape of a
system response curve actually matter that much above even 12 kHz? In fact,
all we know for sure is what we hear, and what we hear even as low as 10 KHz
is actually pretty corrupt. I'm speaking acoustically, of course.
It is well-known in the perceptual coder community that with almost all real
world music, you can throw away *everything* above 16 kHz and nobody is the
wiser. Well they won't be the wiser until they stop listening to the music
and start looking at frequency response curves. They they may have anxiety
attacks. But, its not what they hear that is bugging them. Its what they
think they hear.
dpierce.ca...@gmail.com
> Simple. To accurately recontruct a waveform, you need two samples of the
> highest frequency specificied. Look up Nyquist Theory on you beloved Wiki -
> even Wiki has that right !
Omigod, this is painful!
Look, TWO samples per cycle WILL NOT DO IT.
You MUST have MORE than two samples.
Let me try it a different way
TWO samples per cycle WILL NOT DO IT.
You MUST have MORE than two samples.
dpierce.ca...@gmail.com
> 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.
No, the rolloff is NOT instantaneous, nothing like it.
While maybe steeper than a cassette, it is far from
the hypothetical "brick wall." A theoretically perfect
brick wall filter is not possible to implement in a
practical world.
Rather it merely has to be "good enough," which means
that across the significant audio bandwidth, it has to
affect the audible response in the least possible way,
and as you approach 1/2 the sampling rate, it has to
be far enough down so that any expected images or
aliases are sufficiently attenuated so has to have no
impact.
That means that if you're criteria is to be flat at 20 kHz,
you have a 2.05 kHz transition band available for your
filter, and that provides sufficient margin for many
competent filter implementations.
GregS
Not any waveform. Just a sinewave.
Its interesting aliasing can be seen with nothing to do with time,
but its still a sample, such as with viewing lines through various
filters.
greg
dpierce.ca...@gmail.com
> Not any waveform. Just a sinewave.
No, it's ANY waveform whose components are within
the Nyquist bandwidth (i.e., < 1/2 sample rate) ANY
waveform whatsoever.
Willem
) On Jul 11, 8:24 am, zekfr...@zekfrivolous.com (GregS) wrote:
)> Not any waveform. Just a sinewave.
)
) No, it's ANY waveform whose components are within
) the Nyquist bandwidth (i.e., < 1/2 sample rate) ANY
) waveform whatsoever.
And what are those components ? Sine waves, right ?
Chronic Philharmonic
"geoff" <ge...@nospam-paf.co.nz> wrote in message
news:i7-dna1M87pVpujV...@giganews.com...
The columns on the editor I use are as thick as the dots that represent the
samples. There are a multitude of ways to represent off-screen data in the
UI that does not imply something that is misleading and mathematically
incorrect. I still maintain, if they are going to connect the samples with
lines, the line should reflect the smoothing filter that the sample rate
demands.
Industrial One
[SNIP]
Very eloquent explanation. However, anything spinning over 24 fps will
appear spinning backwards to the human eye anyway, cuz we don't
perceive movement any more precise than that, just like no
neurotypical dude can hear over 20 KHz.
Mark Nelson
>
> And what are those components ? Sine waves, right ?
>
You're stretching it a bit there. A waveform is a waveform. It happens
that we can take an arbitrary waveform and decompose it into a set of
sine waves. But we can also decompose it into other types of
components. It's not like sine waves are some atomic part of any
waveform. It just means that we have built up a body of analysis that
works very well on sine waves, and as a result, decomposing it that
way is convenient.
You can also decompose an arbitrary waveform into wavelets, and then
use different analysis tools. But if this leads to arguments about
whether waveforms are made up of wavelets or sine waves, well, it
shows that we're missing the point.
- Mark
dpierce.ca...@gmail.com
wrote:
No, we do NOT perceive them as "spinning backwards,"
we perceive them as blurred.
There's another phenomenon called "aperture error" or
"aperture filtering" which, in the case of the eye, is due to
the so-called "persistance of vision." It, in effect, "smears"
the continuous input of images.
Now, interestingly enough, as a complete analog
of my wagon-wheel gadanken, how might we prevent
the aliasing of the wagon wheel motion, given that we're
limited to, oh, 25 frames per second?
Well, you "low-pass filter" the wagon wheel, i.e.,
blur its motion, before sampling, so that nothing
makes it through to the "sampler (the shutter"
any faster than 1/2 the shutter or frame rate
That's a little tough to actually accomplish in the
real world. "Aperture filtering" is one way to get
part of the way there: you make your shutter speed
as long as possible. IN the simplest case, it can't
be any longer than 1/25 of a second, which would
have the effect of blurring things enough such that
things goign faster that 1/25 of a second are blurred
wnough that there's nothuing left to alias, but doesn't
solve the problem of what happens between < 12.5
and 25/second.
One might imaging a somewhat more sophisticated'
movie camera that exposes each frame for, say 1/12
second, but interleaves frames such that halway
through frame 1, it starts exposing frames to, and
frame 1 continues unitl halway through frame 2, at
which point frame 3 starts getting esposed, etc.
UnsteadyKen
> Very eloquent explanation. However, anything spinning over 24 fps will
> appear spinning backwards to the human eye anyway, cuz we don't
> perceive movement any more precise than that, just like no
> neurotypical dude can hear over 20 KHz.
We may not "hear" it but the brain registers the presence of
frequencies considerably higher than 20Khz:
http://jn.physiology.org/cgi/content/full/83/6/3548
Similarly we are aware of visual frequencies over 24FPS. Most of us
have come across a faulty fluorescent tube or a computer screen with
a low 50-60hz refresh rate which we sense as an almost subliminal
flicker.
--
Ken
Richard Crowley
> No, we do NOT perceive them as "spinning backwards,"
> we perceive them as blurred.
Then you've never done the experiment youself.
Actually try it and get back to us.
Industrial One
> On Jul 12, 10:58 am, Industrial One <industrial_...@hotmail.com>
> wrote:
>
> > On Jul 10, 6:56 pm, dpierce.cartchunk....@gmail.com wrote:
> > [SNIP]
>
> > Very eloquent explanation. However, anything spinning over 24 fps will
> > appear spinning backwards to the human eye anyway, cuz we don't
> > perceive movement any more precise than that, just like no
> > neurotypical dude can hear over 20 KHz.
>
> No, we do NOT perceive them as "spinning backwards,"
> we perceive them as blurred.
Depends how fast they spin. If enough to capture part of the frame
when you focus your eyes on the oscillation rather than on a fixed
spot at the wheel then it may appear to spin backwards, at least with
me. I believe the limit is... yeah, 'bout 25 fps. If spinning faster,
'course it'll appeared blurred.
> There's another phenomenon called "aperture error" or
> "aperture filtering" which, in the case of the eye, is due to
> the so-called "persistance of vision." It, in effect, "smears"
> the continuous input of images.
>
> Now, interestingly enough, as a complete analog
> of my wagon-wheel gadanken, how might we prevent
> the aliasing of the wagon wheel motion, given that we're
> limited to, oh, 25 frames per second?
>
> Well, you "low-pass filter" the wagon wheel, i.e.,
> blur its motion, before sampling, so that nothing
> makes it through to the "sampler (the shutter"
> any faster than 1/2 the shutter or frame rate
> That's a little tough to actually accomplish in the
> real world. "Aperture filtering" is one way to get
> part of the way there: you make your shutter speed
> as long as possible. IN the simplest case, it can't
> be any longer than 1/25 of a second, which would
> have the effect of blurring things enough such that
> things goign faster that 1/25 of a second are blurred
> wnough that there's nothuing left to alias, but doesn't
> solve the problem of what happens between < 12.5
> and 25/second.
Isn't it already blurred before sampling? No camera captures a
perfectly still image. All got expose times of at least 1/30th a
second, 1/40th 1/50th whatever so if you took a picture of a wheel
spinning faster than that, it'll already be blurred.
> One might imaging a somewhat more sophisticated'
> movie camera that exposes each frame for, say 1/12
> second, but interleaves frames such that halway
> through frame 1, it starts exposing frames to, and
> frame 1 continues unitl halway through frame 2, at
> which point frame 3 starts getting esposed, etc.
You got it.
On Jul 13, 4:24 pm, UnsteadyKen <unsteady...@gmail.com> wrote:
> Industrial One said:
>
> > Very eloquent explanation. However, anything spinning over 24 fps will
> > appear spinning backwards to the human eye anyway, cuz we don't
> > perceive movement any more precise than that, just like no
> > neurotypical dude can hear over 20 KHz.
>
> We may not "hear" it but the brain registers the presence of
> frequencies considerably higher than 20Khz:http://jn.physiology.org/cgi/content/full/83/6/3548
Yeah, it's called binaural beats, you don't hear it, you feel it.. I
once had the priviledge of getting a friend's high-class electronic
playground all to myself for a while, I had this bigass subwoofer
output about 5 Hz and amplified the volume to hardcore maximum. I
didn't hear shit but I swear to god I FELT like shit, like a complete
drop to an emo kid's state -- suicide (I swear to god.) It's real hurl-
inducing too I should note. I wonder what happens at say 500 dB, 1000.
Can some mind-control weapon be built on this idea? Drop an emo bomb
on Iran, say?
I got a shitload of binaural beat files that advertise mind-altering
effects, I only tried S-Angel2 so far (valium) which actually worked.
I gotta try the heroin and microdots ones sometime to see if they
work.
> Similarly we are aware of visual frequencies over 24FPS. Most of us
> have come across a faulty fluorescent tube or a computer screen with
> a low 50-60hz refresh rate which we sense as an almost subliminal
> flicker.
>
> --
> Ken
You're imagining it, take your pills boi.
UnsteadyKen
> Yeah, it's called binaural beats, you don't hear it, you feel it..
It's called infra sound. The US, UK and Russian military investigated
its use as a weapon years ago. It works but can't be projected so it
wasn't practical. The same effect is can cause problems in large
buildings with lift shafts which act as Helmholtz resonators.
It appears to be a very old technique:
http://www.orkneyjar.com/history/tombs/tombacoustics.htm
> once had the priviledge of getting a friend's high-class electronic
> playground all to myself for a while, I had this bigass subwoofer
> output about 5 Hz and amplified the volume to hardcore maximum. I
> didn't hear shit but I swear to god I FELT like shit, like a complete
> drop to an emo kid's state -- suicide (I swear to god.) It's real hurl-
> inducing too I should note. I wonder what happens at say 500 dB, 1000.
> Can some mind-control weapon be built on this idea?
Yes it'scalled shell shock.
> Drop an emo bomb
> on Iran, say?
No I don't say. You can practise on Calgary if you like.
--
Ken
isw
"Richard Crowley" <rcro...@xp7rt.net> wrote:
Are you confusing what happens in the movies ( a sampled data situation
if there ever was one), with what happens in "real life"?
Isaac
Mr.T
"Industrial One" <industr...@hotmail.com> wrote in message
news:f11fa4d4-4eee-437d...@f36g2000hsa.googlegroups.com...
>I wonder what happens at say 500 dB, 1000.
If you mean dB SPL, that is a physical impossibility at normal atmospheric
pressures. From memory a swing to complete vacuum, gives around 180dB SPL,
but of course an asymmetrical swing to greater than twice air pressure can
increase things a bit.
MrT.
dpierce.ca...@gmail.com
> "Industrial One" <industrial_...@hotmail.com> wrote in message
>
> news:f11fa4d4-4eee-437d...@f36g2000hsa.googlegroups.com...
>
> >I wonder what happens at say 500 dB, 1000.
>
> If you mean dB SPL, that is a physical impossibility at normal
> atmospheric pressures. From memory a swing to complete
> vacuum, gives around 180dB SPL,
Actually, this is a common misconception. The low
pressure swing has no need to go to a vacuum for
symmetry. It merely has to match, factor-wise, the
positive swing.
For example, consider a symmetrical pressure swing
where it swings a factor of 2 about atmospheric pressure:
that means at its highest, the pressue is twice that of
atmosphere, and at its loweest,. it's hald that of
atmosphere.
New, where the problem comes is the fact that such
large swings cause nonlinearities because the ideal
gas equation likely no longer holds. Consider:
PV = n R T
which, for small swings in atmospheric pressure
typical with "survivable" (!) sound pressure levels
holds true. THere are some underlyingh assumpitons,
e.g., that based as it is on the kinetic theory of gases,
that the medium consist of essentially point sized
particles that interact through perfectly elastic collisions,
and that the properties of these particles are independent
of temperature, and that, as far as sound is concerned.
These are all true fro small deltas of P, V, and, especially T,
but is increasingly less true as the magnitude of things
start to increase. For example, air molecules behave like
point paritcles at low sound levels, but no so at high
levels. You encounter abrupt discontinuities as energies
start to reach those necessary to effect diassociation of
air molecules, changes in the linearity as more energy is
put into non-translational kinietic ebergy modes, i.e.,
expanding of the molecule, and so on.
But as far as "hitting" the vacuum at the bottom of the
swing: nope. That's not an issue.
Consider one more gedanken: imagine a speaker set in
the wall of a seled room. One could determine that for
all frequecnies whose wavelength is large compared to
the dimensions of the room, the sound pressure level
in the room is a function of the excursion of that speaker.
In other words, assuming we adopt the convention
that excursion of the speaker's diaphragm into the room
is "positive, then since the volume in the room is an
inverse function of the excursion of the diaphragm, and
that pressure is an inverse function volume, then pressure
is a function of diaphragmn excursion.
(for the nit-pickers, whether we are ssuming adiabatic
or isothermal conditions is relatively unimportant at
this point since the end conclusion remains essentially
the same in principle, differing only in magnitude)
Now, let's take you assertion:
"a swing to complete vacuum, gives around
"180dB SPL,
Ignoring whether it's 180 dB or 210 dB or a bazillion
dB, how far sdoes that diaphragm have to move in the
nnegative direction to create a pure vacuum in that room?
(hint, it has to move an infinitie distance).
More importantly, if you want to skip a WHOLE bunch
of physics, how much energy would it take to create that
vacuum? (hint: inifinite).
All this because:
PV = nRt
or,
P = nRT/V
Thus, the relation is a reciprocal one, and when
you plot the relation, you end up with a function whose
graph is asymtotic to the X and Y axes and can never
equal 0.
dpierce.ca...@gmail.com
> <dpierce.cartchunk....@gmail.com> wrote ...
Actually, I have. Any number of times.
Could you elcidtae the conditions under which you
saw the phenomenon?