What's the use of a 192 kHz sample rate?
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Green Xenon [Radium]
Why does DVD-Audio use 192 kHz sample rate? What's the advantage over
44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate?
On average, what is the minimum sample rate for a guy in his early to
mid 20s who likes treble?
I agree there are a small percentage of humans who can hear above 20
kHz. However, DVD-audio uses a sample-rate of 192 kHz which allows a
maximum frequency of 96 kHz. There is no known case of any human being
able to hear sounds nearly as high as 96 kHz. I can agree with 48 kHz
sample rate and even 96 kHz sample-rate [maybe], but 192 kHz is just stupid.
So whats the justification fur using 192 kHz? If you ask me, its just a
total waste of bandwidth and energy. Any proof to the contrary?
Please correct me if I'm wrong but AFAIK, its a waste of time, money,
energy to move to 192 kHz.
Thanks,
Radium
rickman
wrote:
If it really is a waste of time and money to use 192 kHz ADC and DAC,
why do you think they would do it? Don't you think the people
designing DVD equipment understand the economics of consumer
products?
Try to think about it and see if you can come up with a couple of
reasons yourself. I'll be interested in hearing what you think.
Rick
Green Xenon [Radium]
No. I can't think of any reason to use a 192 kHz sample-rate. It is
really overkill. If you think otherwise, the please explain why.
Kevin McMurtrie
Oversampling reduces losses during editing. 192KHz could be the
studio's native format.
It appearing on consumer DVDs might be simply because it falls within
the native bandwidth of a optomechanical mechanism designed for an A/V
stream. A bit of time with a search engine might turn up the answer.
--
Block Google's spam and enjoy Usenet again.
Reply with Google and I won't hear from you.
Randy Yates
> If it really is a waste of time and money to use 192 kHz ADC and DAC,
> why do you think they would do it?
Greed. They think that the general public is dumb enough to buy into
the lie that they really need such a system and would then spend lots of
money repurchasing what they already have.
--
% Randy Yates % "Remember the good old 1980's, when
%% Fuquay-Varina, NC % things were so uncomplicated?"
%%% 919-577-9882 % 'Ticket To The Moon'
%%%% <ya...@ieee.org> % *Time*, Electric Light Orchestra
http://www.digitalsignallabs.com
Randy Yates
> [...]
> On average, what is the minimum sample rate for a guy in his early to
> mid 20s who likes treble?
The curve on p.20 of
http://www.dohc.ie/publications/pdf/hearing.pdf?direct=1
indicates that, even for young adults, sound at <20 kHz is
inaudible. Based on this, a 44.1 kHz sample rate should be ample.
Eeyore
"Green Xenon [Radium]" wrote:
> rickman wrote:
> > "Green Xenon [Radium]" wrote:
> >>
> >> Why does DVD-Audio use 192 kHz sample rate? What's the advantage over
> >> 44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate?
> >>
> >> On average, what is the minimum sample rate for a guy in his early to
> >> mid 20s who likes treble?
> >>
> >> I agree there are a small percentage of humans who can hear above 20
> >> kHz. However, DVD-audio uses a sample-rate of 192 kHz which allows a
> >> maximum frequency of 96 kHz. There is no known case of any human being
> >> able to hear sounds nearly as high as 96 kHz. I can agree with 48 kHz
> >> sample rate and even 96 kHz sample-rate [maybe], but 192 kHz is just stupid.
> >>
> >> So whats the justification fur using 192 kHz? If you ask me, its just a
> >> total waste of bandwidth and energy. Any proof to the contrary?
> >>
> >> Please correct me if I'm wrong but AFAIK, its a waste of time, money,
> >> energy to move to 192 kHz.
> >>
> >
> > If it really is a waste of time and money to use 192 kHz ADC and DAC,
> > why do you think they would do it? Don't you think the people
> > designing DVD equipment understand the economics of consumer
> > products?
> >
> > Try to think about it and see if you can come up with a couple of
> > reasons yourself. I'll be interested in hearing what you think.
>
>
> No. I can't think of any reason to use a 192 kHz sample-rate. It is
> really overkill. If you think otherwise, the please explain why.
Tell me what the point is of making road going cars that can do over 250 mph like
the Bugatti Veyron ? And why would anyone buy one ?
Graham
rickman
> rickman <gnu...@gmail.com> writes:
> > If it really is a waste of time and money to use 192 kHz ADC and DAC,
> > why do you think they would do it?
>
> Greed. They think that the general public is dumb enough to buy into
> the lie that they really need such a system and would then spend lots of
> money repurchasing what they already have.
I'm curious, how do you know what unnamed people are thinking? My
understanding is that regardless of what frequencies acoustic testing
says that people can hear, audiophiles can hear the difference between
many of these "wasteful" features and otherwise adequate audio
systems.
I have known people who worked on professional equipment. The
extremes that they have design in are all audible to the buyers of
such systems. In the audio sections of the equipment they use 15 volt
rails or even higher, just to increase the SNR when the noise floor
can't be lowered anymore. They totally eliminate all digital clocks
from any circuit near the audio section to prevent noise injection.
From what I have seen, they use more extreme measures in high end
audio than is used in sensitive military radio gear which is trying to
get over 140 dB of SNR!
I am not going to try to tell someone else what they can and can't
hear. I know that my hearing has dropped of dramatically to where I
can no longer hear the 15 kHz emitted by TVs and I'm not sure I can
hear the high notes on a piano. When I press the keys on the right, I
hear more of a click than a ping (maybe it's the piano)! But that
doesn't mean that there aren't others who can hear the distortion
created by the anti-alias filters used when the ADC and DAC run at
44.1 kHz.
Don Pearce
wrote:
>On May 3, 3:28 am, Randy Yates <ya...@ieee.org> wrote:
>> rickman <gnu...@gmail.com> writes:
>> > If it really is a waste of time and money to use 192 kHz ADC and DAC,
>> > why do you think they would do it?
>>
>> Greed. They think that the general public is dumb enough to buy into
>> the lie that they really need such a system and would then spend lots of
>> money repurchasing what they already have.
>
>I'm curious, how do you know what unnamed people are thinking? My
>understanding is that regardless of what frequencies acoustic testing
>says that people can hear, audiophiles can hear the difference between
>many of these "wasteful" features and otherwise adequate audio
>systems.
>
Utter nonsense - unless of course you can cite some proper tests.
>I have known people who worked on professional equipment. The
>extremes that they have design in are all audible to the buyers of
>such systems. In the audio sections of the equipment they use 15 volt
>rails or even higher, just to increase the SNR when the noise floor
>can't be lowered anymore. They totally eliminate all digital clocks
>from any circuit near the audio section to prevent noise injection.
>From what I have seen, they use more extreme measures in high end
>audio than is used in sensitive military radio gear which is trying to
>get over 140 dB of SNR!
>
More nonsense. I have worked on many radio systems, both military and
civil, and a typical target SNR for these radios is in the region of 6
to 10dB.
>I am not going to try to tell someone else what they can and can't
>hear. I know that my hearing has dropped of dramatically to where I
>can no longer hear the 15 kHz emitted by TVs and I'm not sure I can
>hear the high notes on a piano. When I press the keys on the right, I
>hear more of a click than a ping (maybe it's the piano)! But that
>doesn't mean that there aren't others who can hear the distortion
>created by the anti-alias filters used when the ADC and DAC run at
>44.1 kHz.
Doesn't mean they can either. That isn't a piece of logic that
commutes.
d
--
Pearce Consulting
http://www.pearce.uk.com
Randy Yates
> [...]
> When I press the keys on the right, I
> hear more of a click than a ping (maybe it's the piano)!
Then you can no longer hear up to 4186 Hz.
http://en.wikipedia.org/wiki/Piano_key_frequencies
You are free to think what you want about hearing. I choose to side with
objective measurements verifying again and again over several decades
the same conclusions rather than a few crackpot audiophools that claim
they're different without any supporting objective evidence.
And even if one or two actually could hear beyond 20 kHz, they're the
Robert Wadlow's of the audio world - should we start building houses
with 10-foot ceilings because 1 out of a billion will be over 8 feet
tall?
--
% Randy Yates % "How's life on earth?
%% Fuquay-Varina, NC % ... What is it worth?"
%%% 919-577-9882 % 'Mission (A World Record)',
%%%% <ya...@ieee.org> % *A New World Record*, ELO
http://www.digitalsignallabs.com
rickman
This thread is one that could be thought provoking. But I would like
to see *you* do some of the thinking. What is the difference between
the way a player is designed using a 44.1 (or 48) kHz sample rate and
a 192 kHz sample rate? What does that difference do to the analog
signal that is produced? How might that difference sound to a
listener?
This is not rocket science. The differences may not be audible to
you, but they exist and they are well understood. I'll give you a
hint; do you know what phase distortion is?
Also, I would like you to explain what is wasteful about a 192 kHz
sample rate. How much money does it cost to use 192 kHz instead of 96
kHz or 44.1 kHz. What time is involved? How much extra energy does
the higher sample rate use? Are you aware that there are virtually
infinite amounts of bandwidth not being used every second? What
difference does it make if DVDs underutilize a bit more?
I am working on a circuit that uses a 192 kHz CODEC to process 1 kHz
signals. Of course I am not using it at 192 kHz, so am I wasting
bandwidth still? Or by using an 8 kHz sample rate, am I conserving
bandwidth and deserve recognition? I like the idea of being "pink" (as
in noise) by conserving precious bandwidth. Actually, I can't wait to
get the thing out on the open road and open it up! I want to put the
pedal to the metal and sample at the full 192 kHz to see just what
sort of analog bandwidth these CODECs really have! I'm not actually
sure they will produce higher than about 40 kHz at the analog output.
I'll have to hack away the 1 pole filter I added to the input and
output. It limits the frequency range to a paltry 30 kHz or so. I
may not hear the difference, but my scope can "see" it!
BTW, if you aren't looking at your monitor or you aren't sitting close
enough to see every pixel, are you wasting bandwidth on your video
signal? Actually, that is an excellent analogy. If you limit the
video bandwidth to half the pixel rate, what difference will you see
on a CRT display?
Finally, why do you see the advantage of using 96 kHz sample rate and
not 192 kHz? Isn't 96 kHz wasting bandwidth?
Piergiorgio Sartor
> The curve on p.20 of
>
> http://www.dohc.ie/publications/pdf/hearing.pdf?direct=1
>
> indicates that, even for young adults, sound at <20 kHz is
> inaudible. Based on this, a 44.1 kHz sample rate should be ample.
The problem might be that a "sound" is not only
perceived with ears, but with the full body.
One specific issue are the sudden transitions of
pitches that real world instruments can create,
and, possibly, are picked up by different means.
For example, the sound waves going through the
skull, brain tissue and then to ears (from inside)
are "distorted" and, maybe, remodulated, so that
a person *could* receive "signals" (from real
instruments), which otherwise would be wiped out
by sampling low-pass filter, in case of 44.1KHz
sampling rate.
That's the theory.
Is this true? I don't know, maybe, maybe not.
There are people swearing this is the case, others
say that's nonsense.
There are people citing Dolby technologies taking
into account these alternate perception paths, but
I could not find any real reference.
Nevertheless, one thing is sure (and proven) sound
waves are not only "eared", they're generally
perceived by the full body, but the studies I know
always refer to bass and not to ultrasounds.
bye,
--
piergiorgio
rajesh
<piergiorgio.sartor.this.should.not.be.u...@nexgo.REMOVETHIS.de>
wrote:
I agree that humans cant hear above 20k rate
Its not always about what you hear or percieve with your body.
Its also about how you store data.
here is an simplified analogy.
say you need 44.1k samples per second to hear properly. If the disk
is corrupted with scrathes
and 1 samples in his region are lost your sound is distorted or
lost for that period of time.
Now if there are 196k samples even if (196/44.1) samples are lost
there is no difference to what you
hear.
DVD's come wih high density of data due to this they are highly
vulnerable to scratches this can be avoided
with better waveform matching achieved by high sampling rate.
rajesh
I am ot sure about the math, its worh doing again. This is one
argument in favour of
high sample rate.
Ethan Winer
> Greed. They think that the general public is dumb enough to buy into the
> lie that they really need such a system and would then spend lots of money
> repurchasing what they already have.
LOL, you nailed it man. Same with music, where they hope you'll buy all the
same titles again to play on your new unnecessary hardware.
Besides the obvious waste, as Radium pointed out, this nonsense has been
discredited fully:
Audio Critic summary of the AES hi-res fallacy article:
http://theaudiocritic.com/blog/index.php?op=ViewArticle&articleId=41&blogId=1
More from the authors themselves:
http://www.bostonaudiosociety.org/explanation.htm
Paul Lehrman commenting in Mix magazine:
http://mixonline.com/recording/mixing/audio_emperors_new_sampling/index1.html
Nobody can hear, or perceive, or be influenced by ultrasonic content, even
if they think they can. Here's my best explanation for why people sometimes
report hearing differences even when none can possibly exist:
http://www.ethanwiner.com/believe.html
> And even if one or two actually could hear beyond 20 kHz, they're the
> Robert Wadlow's of the audio world - should we start building houses with
> 10-foot ceilings because 1 out of a billion will be over 8 feet tall?
More to the point, even if a few people really can just barely detect when
frequencies above 20 KHz are removed, who cares? Just because someone can
perceive 21 KHz doesn't mean that music they hear must contain those
frequencies to be satisfying.
--Ethan
rickman
> On Sat, 3 May 2008 05:11:53 -0700 (PDT), rickman <gnu...@gmail.com>
> wrote:
>
> >On May 3, 3:28 am, Randy Yates <ya...@ieee.org> wrote:
> >> rickman <gnu...@gmail.com> writes:
> >> > If it really is a waste of time and money to use 192 kHz ADC and DAC,
> >> > why do you think they would do it?
>
> >> Greed. They think that the general public is dumb enough to buy into
> >> the lie that they really need such a system and would then spend lots of
> >> money repurchasing what they already have.
>
> >I'm curious, how do you know what unnamed people are thinking? My
> >understanding is that regardless of what frequencies acoustic testing
> >says that people can hear, audiophiles can hear the difference between
> >many of these "wasteful" features and otherwise adequate audio
> >systems.
>
> Utter nonsense - unless of course you can cite some proper tests.
And what do you base this statement on? I don't have any "proper"
studies. I am referring to a conversation with a friend who worked in
the field. He couldn't hear the difference, but his customers could.
If they came out with a product that used a "lesser" technique,
without *knowing* what technology was behind it they would reject the
system as not being good enough. If they didn't know anything about
the methods, they only had their ears to judge the equipment. Since
they are paying serious bucks (>$100,000 some 20 years ago) for these
systems, they have *NO* reason to buy racks of gear that is any better
than a lower price system. BTW, this was professional equipment, not
the home stuff with oxygen free speaker wires and such.
You can poo-poo this sort of evaluation. But that doesn't make you
right. Do you have any "proof" that no one can hear the difference?
Do you even know what the differences are that I was talking about?
> >I have known people who worked on professional equipment. The
> >extremes that they have design in are all audible to the buyers of
> >such systems. In the audio sections of the equipment they use 15 volt
> >rails or even higher, just to increase the SNR when the noise floor
> >can't be lowered anymore. They totally eliminate all digital clocks
> >from any circuit near the audio section to prevent noise injection.
> >From what I have seen, they use more extreme measures in high end
> >audio than is used in sensitive military radio gear which is trying to
> >get over 140 dB of SNR!
>
> More nonsense. I have worked on many radio systems, both military and
> civil, and a typical target SNR for these radios is in the region of 6
> to 10dB.
My bad, I used the wrong term, it should have been 140 dB signal
strength. Yes, I need to indicate what the reference is, but I don't
recall if it was dBmW or dBW, a 30 dB difference.
> >I am not going to try to tell someone else what they can and can't
> >hear. I know that my hearing has dropped of dramatically to where I
> >can no longer hear the 15 kHz emitted by TVs and I'm not sure I can
> >hear the high notes on a piano. When I press the keys on the right, I
> >hear more of a click than a ping (maybe it's the piano)! But that
> >doesn't mean that there aren't others who can hear the distortion
> >created by the anti-alias filters used when the ADC and DAC run at
> >44.1 kHz.
>
> Doesn't mean they can either. That isn't a piece of logic that
> commutes.
I didn't say it proves that others can. What logic are you talking
about? I am simply saying that you shouldn't judge what others can
perceive by what you can or even what the general public can according
to "proper tests".
I'm not trying to "prove" anything. I am presenting information which
you can consider and believe or can ignore. But you can't say my
statements are false unless you have some information to "prove" they
are. Human hearing is not a microphone connected to an amplifier. It
is a very complex process which even includes the brain and we
certainly don't understand it completely.
rickman
> rickman <gnu...@gmail.com> writes:
> > [...]
> > When I press the keys on the right, I
> > hear more of a click than a ping (maybe it's the piano)!
>
> Then you can no longer hear up to 4186 Hz.
>
> http://en.wikipedia.org/wiki/Piano_key_frequencies
Like I said, it may be the piano that doesn't really produce much of a
ping on the highest notes. But all of us have some hearing loss as we
get older. The point is that you can't judge how others hear based on
what we can hear.
> You are free to think what you want about hearing. I choose to side with
> objective measurements verifying again and again over several decades
> the same conclusions rather than a few crackpot audiophools that claim
> they're different without any supporting objective evidence.
>
> And even if one or two actually could hear beyond 20 kHz, they're the
> Robert Wadlow's of the audio world - should we start building houses
> with 10-foot ceilings because 1 out of a billion will be over 8 feet
> tall?
I never said that *anyone* can hear sounds above 20 kHz. I said that
there are people who can hear the difference between systems that use
higher sample rates than 48 kHz. The real issue is about the
electronics and how they are measured *compared* to human hearing.
Distortion is the real issue. Filters introduce distortion and some
people can hear the distortion produced by converters running at 48
kHz.
I suppose it is possible that my friend was mislead (and therefor
myself). I have no first hand knowledge of the sound tests. But he
is not a person to believe in rubbish. In fact, he is a person who is
very intellectually critical. This is one of the few times he could
not dismiss claims that run counter to the science.
Oh, about the Robert Wadlow's of the acoustic world... if lumber in 10
foot lengths were the same price as 8 foot lengths, why *not* build
homes with 10 foot ceilings? At one time 9 foot ceilings were the
norm. What is the delta in price, power or anything else going from
48 kHz to 196 kHz sample rates??? My present design is using a CODEC
capable of 192 kHz sample rate even though I am running it at 8 kHz.
It is actually a bit cheaper than a similar chip from the same maker
that only runs at 96 kHz. The power consumption is only a few mW
higher and I have no idea of what the OP meant by the waste of
"time"...
Rick
Vladimir Vassilevsky
rickman wrote:
>>Utter nonsense - unless of course you can cite some proper tests.
>
> And what do you base this statement on?
The ultimate reason for the audio systems is making the people happy. If
someone is happy because of 192kHz sample rate, and willing to pay for
that, then why do you need to proove anything? Heck, if someone orders a
192MHz audio system, it would be my pleasure to do this project.
Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com
Steve Underwood
>
>
> rickman wrote:
>
>
>>> Utter nonsense - unless of course you can cite some proper tests.
>>
>> And what do you base this statement on?
>
> The ultimate reason for the audio systems is making the people happy. If
> someone is happy because of 192kHz sample rate, and willing to pay for
> that, then why do you need to proove anything? Heck, if someone orders a
> 192MHz audio system, it would be my pleasure to do this project.
Quite right, too. If engineers only built what people actually need, you
and I would probably be looking for another occupation.
Steve
rickman
On May 3, 10:19 am, "Ethan Winer" <ethanw at ethanwiner dot com>
wrote:
> Randy,
>
> > Greed. They think that the general public is dumb enough to buy into the
> > lie that they really need such a system and would then spend lots of money
> > repurchasing what they already have.
>
> LOL, you nailed it man. Same with music, where they hope you'll buy all the
> same titles again to play on your new unnecessary hardware.
>
> Besides the obvious waste, as Radium pointed out, this nonsense has been
> discredited fully:
>
> Audio Critic summary of the AES hi-res fallacy article:http://theaudiocritic.com/blog/index.php?op=ViewArticle&articleId=41&...
This seems to indicate that the tests do show the higher formats to be
equivalent to CDs. I assume that when they refer to a "16-bit/44.1-
kHz A/D/A processor" they mean the are digitizing and then restoring
to analog a signal from a higher end system. Assuming that the tests
were done correctly, and I have no reason to believe they weren't,
this seems pretty convincing.
> More from the authors themselves:http://www.bostonaudiosociety.org/explanation.htm
Maybe I just don't understand this report, but I see sections where
they claim that the blind tests showed 100% correct results??!!!
Doesn't that indicate that the listener *could* tell the difference?
Unfortunately the writeup is very lengthy in the description of the
systems and very, very terse in the description of what they are
actually measuring... Correct me if I am misinterpreting their
results.
> Paul Lehrman commenting in Mix magazine:http://mixonline.com/recording/mixing/audio_emperors_new_sampling/ind...
This is less clear. He is just commenting on the tests cited in the
first article. The actual test report costs $20 from AES.
> Nobody can hear, or perceive, or be influenced by ultrasonic content, even
> if they think they can. Here's my best explanation for why people sometimes
> report hearing differences even when none can possibly exist:
The question is *not* about ultrasonic content. The question is about
recording and playback systems using different sample rates.
> http://www.ethanwiner.com/believe.html
>
> > And even if one or two actually could hear beyond 20 kHz, they're the
> > Robert Wadlow's of the audio world - should we start building houses with
> > 10-foot ceilings because 1 out of a billion will be over 8 feet tall?
>
> More to the point, even if a few people really can just barely detect when
> frequencies above 20 KHz are removed, who cares? Just because someone can
> perceive 21 KHz doesn't mean that music they hear must contain those
> frequencies to be satisfying.
You are free to buy any system you want. Why do you care what other
people use?
rickman
> "Green Xenon [Radium]" <gluceg...@excite.com> writes:
>
> > [...]
> > On average, what is the minimum sample rate for a guy in his early to
> > mid 20s who likes treble?
>
> The curve on p.20 of
>
> http://www.dohc.ie/publications/pdf/hearing.pdf?direct=1
>
> indicates that, even for young adults, sound at <20 kHz is
> inaudible. Based on this, a 44.1 kHz sample rate should be ample.
I think I am spending too much time on this, but your conclusion does
not follow from the data. It's that simple. The issue is whether the
lower sample rates work as well as higher sample rates in recording
and playback systems. Your cursory analysis of human hearing
frequency response does not indicate anything in that regard.
Don Pearce
wrote:
That is a truly idiot comment. To prove that nobody could hear it I
would have to test everybody in the world for the rest of time with
every possible type of music - and even then there is no guarantee
that the next thing I tried wouldn't be the one that revealed the
difference. All you need to do is produce one person who can reliably
hear the difference with their choice of material and the job is done.
And no, anecdote won't do. This business is full of bullshit anecdotes
- mostly involving wives hearing the difference from the kitchen even
when they didn't know what had been done.
>
>> >I have known people who worked on professional equipment. The
>> >extremes that they have design in are all audible to the buyers of
>> >such systems. In the audio sections of the equipment they use 15 volt
>> >rails or even higher, just to increase the SNR when the noise floor
>> >can't be lowered anymore. They totally eliminate all digital clocks
>> >from any circuit near the audio section to prevent noise injection.
>> >From what I have seen, they use more extreme measures in high end
>> >audio than is used in sensitive military radio gear which is trying to
>> >get over 140 dB of SNR!
>>
>> More nonsense. I have worked on many radio systems, both military and
>> civil, and a typical target SNR for these radios is in the region of 6
>> to 10dB.
>
>My bad, I used the wrong term, it should have been 140 dB signal
>strength. Yes, I need to indicate what the reference is, but I don't
>recall if it was dBmW or dBW, a 30 dB difference.
>
So a meaningless number then?
>
>> >I am not going to try to tell someone else what they can and can't
>> >hear. I know that my hearing has dropped of dramatically to where I
>> >can no longer hear the 15 kHz emitted by TVs and I'm not sure I can
>> >hear the high notes on a piano. When I press the keys on the right, I
>> >hear more of a click than a ping (maybe it's the piano)! But that
>> >doesn't mean that there aren't others who can hear the distortion
>> >created by the anti-alias filters used when the ADC and DAC run at
>> >44.1 kHz.
>>
>> Doesn't mean they can either. That isn't a piece of logic that
>> commutes.
>
>I didn't say it proves that others can. What logic are you talking
>about? I am simply saying that you shouldn't judge what others can
>perceive by what you can or even what the general public can according
>to "proper tests".
>
It isn't a question of "others" - one is all it takes, and I can
assure you that nobody has produced him yet.
>I'm not trying to "prove" anything. I am presenting information which
>you can consider and believe or can ignore. But you can't say my
>statements are false unless you have some information to "prove" they
>are. Human hearing is not a microphone connected to an amplifier. It
>is a very complex process which even includes the brain and we
>certainly don't understand it completely.
No, it isn't information - it is urban myth and anecdote.
rickman
I thought we were having a conversation. But I don't appreciate being
called names. Would you speak to me this way if I were standing in
front of you? Either way you come across as being rude.
Rick
Steve Underwood
The 22kHz scanning monitors I used to use drove me crazy right up until
my late 30s. Then, the monitors started scanning higher, and my ears
were getting older. I'm not clear which one caused me to be free of the
whistling first. :-)
I think a lot of the arguments about sampling rate muddle two things up
- which, of course, the people who stand to profit from snake oil love
very much - the rate at which you convert, and the rate at which you
transmit or store.
I think there is great merit in sampling at 192k/s. These days a 192k/s
24 bit stereo DAC offers excellent noise and distortion specs, and costs
20 cents. Such a high sample rate really makes the analogue filtering a
lot easier. A 192k/s ADC is not much more expensive (a difference
probably driven more by volume than complexity).
Actually transmitting and storing such sample rates makes no sense at
all. 44.1k/s was a bit marginal, when you allow for the impracticality
of the filters getting really close to 0.5fs. However, 48k/s should be
good enough for any practical purpose.
For people who say supersonic sound can't play a part in a listening
experience, trying being in a room with a high intensity of supersonic
energy. Under some conditions (I'm not clear which) you can sense it,
even though you can't hear it. It actually feels like something loud
that you can't hear is going on. Its a very odd feeling. That said, I've
never found any evidence that this plays a part in any musical
experience. I see no reason to try to capture that energy in a
recording, unless you feel your dog should enjoy a greater musical
experience.
Steve
Dave Platt
Green Xenon [Radium] <gluc...@excite.com> wrote:
>
>Why does DVD-Audio use 192 kHz sample rate? What's the advantage over
>44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate?
>
>On average, what is the minimum sample rate for a guy in his early to
>mid 20s who likes treble?
In my personal opinion, a rate of 192 kilosamples/second (allowing a
passband of DC to 96 kHz) is indeed overkill as an audio delivery
standard.
There may be _some_ justification for it, as it eliminates the need to
place the knee of the anti-aliasing filter anywhere near the range of
frequencies that one _can_ hear. One of the criticisms made against
CD is that the sharp filtering which must be done at around 20 kHz can
cause artifacts which may be audible to some listeners, either due to
"pre-ringing" (with a symmetric FIR low-lass filter) or a frequency-
dependent delay and "smearing" of transients (with an IIR filter).
These effects can be moved up to higher frequencies, and prevented
from having effects in the human hearing passband, by increasing the
sampling rate.
My own personal guess is that a rate of 96 ksamples/second is probably
high enough to move any phase-affecting artifacts up well out of the
human hearing range, and that there are few if any benefits to going
to a rate higher than this.
>So whats the justification fur using 192 kHz? If you ask me, its just a
>total waste of bandwidth and energy.
The likely reasons are twofold:
- The "numbers game". "More is better", at least in advertising
literature. The consumer-electronics company needed a New Big Thing
to sell a new generation of products (everybody already had CD) and
the media companies needed a New Better Delivery Platform to
convince everybody to buy new editions of music they already own
("... have to buy the White Album again!" per Men In Black).
Having an extremely high sample rate makes the new system sound
even more attractive to the naive consumer.
- The "why not?" issue. The DVD storage technology has lots of
space and bandwidth available, as it was designed for video.
Storage space on the discs is not an issue... one could use an even
higher rate than 192k, store a 90-minute album per disc, and still
have space left over.
>Please correct me if I'm wrong but AFAIK, its a waste of time, money,
>energy to move to 192 kHz.
Given that DVD is used as the storage system, the costs of using such
a high data rate are negligible.
--
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!
Don Pearce
wrote:
>On May 3, 12:14 pm, nos...@nospam.com (Don Pearce) wrote:
I'll have a reasonable conversation with anyone who can make
reasonable points. Your demand that I prove a negative fell well
outside that area. Talk to me like an idiot, and I will treat you that
way. Your choice.
Scott Seidman
news:481becfe$0$5141$4c36...@roadrunner.com:
> Hi:
>
> Why does DVD-Audio use 192 kHz sample rate? What's the advantage over
> 44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate?
>
> On average, what is the minimum sample rate for a guy in his early to
> mid 20s who likes treble?
>
> I agree there are a small percentage of humans who can hear above 20
> kHz. However, DVD-audio uses a sample-rate of 192 kHz which allows a
> maximum frequency of 96 kHz. There is no known case of any human being
> able to hear sounds nearly as high as 96 kHz. I can agree with 48 kHz
> sample rate and even 96 kHz sample-rate [maybe], but 192 kHz is just
> stupid.
>
> So whats the justification fur using 192 kHz? If you ask me, its just
> a total waste of bandwidth and energy. Any proof to the contrary?
>
> Please correct me if I'm wrong but AFAIK, its a waste of time, money,
> energy to move to 192 kHz.
>
>
> Thanks,
>
> Radium
>
Is this a sampling rate or an oversampling rate?
--
Scott
Reverse name to reply
vlad
wrote:
I guess, you are in business of separating fool from his money.
vlad
Oli Charlesworth
>
> There may be _some_ justification for it, as it eliminates the need to
> place the knee of the anti-aliasing filter anywhere near the range of
> frequencies that one _can_ hear. One of the criticisms made against
> CD is that the sharp filtering which must be done at around 20 kHz can
> cause artifacts which may be audible to some listeners, either due to
> "pre-ringing" (with a symmetric FIR low-lass filter) or a frequency-
> dependent delay and "smearing" of transients (with an IIR filter).
>
> These effects can be moved up to higher frequencies, and prevented
> from having effects in the human hearing passband, by increasing the
> sampling rate.
Indeed. But this problem is largely solved by oversampling at the
DAC, without requiring an increased sample rate at the storage-medium
level.
--
Oli
Vladimir Vassilevsky
The only real needs are the air, the food and the water. Everything else
is a luxury, including sex. The ultimate goal for building anything is
making somebody happier. If this goal is accomplished by one way or
another, then the project makes sense.
robert bristow-johnson
> I am simply saying that you shouldn't judge what others can
> perceive by what you can or even what the general public can according
> to "proper tests".
what *should* we base it on? what the Monster Cable people tell us
that they can hear?
by "proper tests", i am convinced what they mean are truly blind tests
where a sufficient number of subjects, with "good" hearing, are called
on to say if they can hear a difference between two possibly different
sounds or not. rather than "ABX testing", i think it should be "AB
testing" where the subject hears groups of two sounds, and for each
group the only question the subject must answer is if he/she thinks
the sounds are different or the same (*not* which one sounds "better"
or "less distorted or noisy", etc.). the test would include an equal
number of identical pairs so that we can unbias it from the subject's
pre-conceived biases. for each subject you would subtract the number
of false positives from the number of true positives and also subtract
the number of false negatives from the number of true negatives.
now, long ago, when DSD was just coming out, i participated in
something like this that an audio guy (now an author) named Bob Katz
did. this was before Pro Tools HD and they were using some expensive
system from a company called Sonic Solutions that could do 192 kHz.
they recorded at 192 kHz some test sounds (including some high
frequency percussive sounds like castenets, cabasas, cymbals) along
with synthesized bandlimited (to 96 kHz) tones of all sorts of
waveshapes at a variety of frequencies from below 10 kHz up to 90
kHz. i don't remember all of the test sounds, but they made sure that
most, if not all, had content well above 30 kHz. the subject listened
to those sounds in two different forms, but both with a 192 kHz
playback rate. one form was the raw recorded sound, the other was
processed through a phase-linear FIR filter with a lot of taps (i
thought it was around 300 something taps, it was not a real-time FIR
filter but proceessed one sound file into another) that was flat to
within 0.01 dB up to 20 kHz, had a smooth transition band from 20 to
22 kHz, and then was down by more than 130 dB for 22 kHz to 96 kHz.
we padded the beginning an end of the original sound file with zeros,
of half the length of the FIR on both ends and the non-real-time FIR
was not "causal" and had a delay of 0 (so it started responding before
the first non-zero samples) and the filtered sound file was lined up
in time with the original.
now this isn't what Bob Katz did, but it is what i wished he did:
called the original sound "A", and the LPFed sound "B". so sometimes
the subject hears AA, sometimes AB, sometimes BA, and sometimes BB
(all four permutations exist in equal quantity) and for each pair, the
subject simply has to say if they think the sounds are the same or
different. every time a subject says that AA or BB sound different,
we count that as a false positive and subtract that count from the
number of times the subject says that AB or BA sound different.
likewise for false negatives.
now Bob didn't do that, but he did something like it (i think it was
ABX) and there was no statistically measured difference. people could
simply not reliably tell if the stuff above 20 kHz was removed or
not. they could not tell at all. no one could.
now, if they cannot tell if the content existed above 22 kHz or not,
is there a need to have it there in storage or in transmission? if
there is no need to have it there, and it is removed, what does the
sampling theorem tell us regarding sufficient sampling rate?
> I'm not trying to "prove" anything. I am presenting information which
> you can consider and believe or can ignore. But you can't say my
> statements are false unless you have some information to "prove" they
> are. Human hearing is not a microphone connected to an amplifier. It
> is a very complex process which even includes the brain and we
> certainly don't understand it completely.
but we can't hear anything above 20 kHz. even with percussive sounds
with sharp attacks. and if we cannot hear anything above 20 kHz, then
40.0001 kHz sampling rate can store all of the information we need.
for practical reconstruction purposes, 44.1 and 48 kHz are sufficient.
now, in *processing* sounds with some nasty non-linearities in the
process, it very well may be necessary to upsample to 192 kHz or
higher to do that non-linear processing, and when it is done, LPF to
20 kHz and downsample back to 48 kHz. but, except for experimental
purposes, 192 kHz storage or transimssion is not necessary.
r b-j
Ken
<r...@audioimagination.com> wrote:
> but we can't hear anything above 20 kHz. even with percussive sounds
> with sharp attacks. and if we cannot hear anything above 20 kHz, then
> 40.0001 kHz sampling rate can store all of the information we need.
> for practical reconstruction purposes, 44.1 and 48 kHz are sufficient.
No. When I was 35 years I could hear up to 24 kHz.
Now I'm 54 years and can easy hear 19 kHz (haven't tested higher).
Nils
>
> For people who say supersonic sound can't play a part in a listening
> experience, trying being in a room with a high intensity of supersonic
> energy. Under some conditions (I'm not clear which) you can sense it,
> even though you can't hear it. It actually feels like something loud
> that you can't hear is going on. Its a very odd feeling. That said, I've
> never found any evidence that this plays a part in any musical
> experience. I see no reason to try to capture that energy in a
> recording, unless you feel your dog should enjoy a greater musical
> experience.
I remember that there was some kind of ultrasound sound-gun to play with
on the expo2000 in hannover/germany. It modulated AF-sound onto an
ultrasound signal and transmitted it in a focused sound-beam.
You could point the beam at people from a far distance, say something
into a microphone and the sound demodulated right inside their bones. It
was very direct and you had to be somewhere in a very narrow funnel to
notice it.
I can tell you: It's a very scary experience because your brain
localizes these sounds somewhere in your own body.
robert bristow-johnson
well, good for you. still would like to see how you would do in such
a blind test.
r b-j
Green Xenon [Radium]
> In article <481becfe$0$5141$4c36...@roadrunner.com>,
> "Green Xenon [Radium]" <gluc...@excite.com> wrote:
>
>> Hi:
>>
>> Why does DVD-Audio use 192 kHz sample rate? What's the advantage over
>> 44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate?
>>
>> On average, what is the minimum sample rate for a guy in his early to
>> mid 20s who likes treble?
>>
>> I agree there are a small percentage of humans who can hear above 20
>> kHz. However, DVD-audio uses a sample-rate of 192 kHz which allows a
>> maximum frequency of 96 kHz. There is no known case of any human being
>> able to hear sounds nearly as high as 96 kHz. I can agree with 48 kHz
>> sample rate and even 96 kHz sample-rate [maybe], but 192 kHz is just stupid.
>>
>> So whats the justification fur using 192 kHz? If you ask me, its just a
>> total waste of bandwidth and energy. Any proof to the contrary?
>>
>> Please correct me if I'm wrong but AFAIK, its a waste of time, money,
>> energy to move to 192 kHz.
>>
>>
>> Thanks,
>>
>> Radium
>
> studio's native format.
>
> It appearing on consumer DVDs might be simply because it falls within
> the native bandwidth of a optomechanical mechanism designed for an A/V
> stream. A bit of time with a search engine might turn up the answer.
>
Shouldn't a 96 kHz sample-rate suffice for that?
rickman
And how exactly do you over sample at the DAC without a filter?
Rick
dpierce.ca...@gmail.com
And where exactly is he claiming otherwise?
Oversampling DACs have been the rule for well over two
decades.
dpierce.ca...@gmail.com
wrote:
> >> Please correct me if I'm wrong
Why would anyone think that would work now?
>>> but AFAIK, its a waste of time, money,
>>> energy to move to 192 kHz.
Rick Lyons
wrote:
>rickman <gnu...@gmail.com> writes:
>
>> If it really is a waste of time and money to use 192 kHz ADC and DAC,
>> why do you think they would do it?
>
>Greed. They think that the general public is dumb enough to buy into
>the lie that they really need such a system and would then spend lots of
>money repurchasing what they already have.
Hi Randy,
you remind me of the transistor radios when I was
kid (back when the air was clean, and sex was dirty).
If a transistor radio manufacturer could claim that
their radio had had more transistors than their competition,
then that was strong "selling point". As such,
some transistor radio manufacturers were using transitors
in place of the diodes needed in AM demodulation.
So instead of having four transistors and one diode,
those manufacturers could claim "5-transistor performance",
in the hope of increasing sales. Ha ha.
See Ya,
[-Rick-]
Dave Platt
Rick Lyons <R.Lyons@_BOGUS_ieee.org> wrote:
>Hi Randy,
> you remind me of the transistor radios when I was
>kid (back when the air was clean, and sex was dirty).
>If a transistor radio manufacturer could claim that
>their radio had had more transistors than their competition,
>then that was strong "selling point". As such,
>some transistor radio manufacturers were using transitors
>in place of the diodes needed in AM demodulation.
>So instead of having four transistors and one diode,
>those manufacturers could claim "5-transistor performance",
>in the hope of increasing sales. Ha ha.
I've read that there were some "7-transistor" radios, in which one or
two of the transistors were soldered to unconnected pads on the board.
They had no function at all, and they were often "floor sweeping"
parts known to be defective... but they _were_ transistors and were
present in the radio, and so the radio could be advertised (legally if
not all that ethically) as a "7-transistor" model.
Green Xenon [Radium]
Unfortunately, no. I've done as my of my own research on this subject
and still don't understand why a 96 kHz sample-rate is inadequate. I
still don't understand why a 192 kHz sample-rate is at all necessary.
robert bristow-johnson
> In article <96fq141o7bh82pho575op2q1upefkq6...@4ax.com>,
> Rick Lyons <R.Lyons@_BOGUS_ieee.org> wrote:
>
> >Hi Randy,
> > you remind me of the transistor radios when I was
> >kid (back when the air was clean, and sex was dirty).
> >If a transistor radio manufacturer could claim that
> >their radio had had more transistors than their competition,
> >then that was strong "selling point". As such,
> >some transistor radio manufacturers were using transitors
> >in place of the diodes needed in AM demodulation.
> >So instead of having four transistors and one diode,
> >those manufacturers could claim "5-transistor performance",
> >in the hope of increasing sales. Ha ha.
>
> I've read that there were some "7-transistor" radios, in which one or
> two of the transistors were soldered to unconnected pads on the board.
> They had no function at all, and they were often "floor sweeping"
> parts known to be defective... but they _were_ transistors and were
> present in the radio, and so the radio could be advertised (legally if
> not all that ethically) as a "7-transistor" model.
i thought i remembered even "10-transistor" radios. i didn't know
this about the bogus addition of transistors (when i was older and
reflected on it, i just thought that it meant more stages in the RF,
IF, and AF signal chain or the use of some push-pull pairs.
if they were so bad as to solder in unused transistors, they should
have been clever enough to route the pads into the circuit board in
some non-functional way (leaving the transistor shorted and hanging
offa something, so it at least "looked" like it was being used for
something.
it's sorta like when someone slips in superfluous instructions in
their code so that they can later prove that it wasn't developed
independently if they find someone else selling a product with their
code in it.
r b-j
Randy Yates
I never knew that! I remember the "N-transistor radio" slogans of the
mid to late 60s (if I recall correctly). Like Robert, I distinctly
remember "10-transistor" radios.
Just to go totally off-topic, this also reminds me of a couple of my
early electronics love affairs. Back in circa 1966 or 67 (about the time
the Beatles' "White Album" came out) I got a really cool device for
Christmas (or birthday - they're 8 days apart): an AM/FM radio AND a
really neat miniature reel-to-reel mag tape recorder! Oh how I loved
that radio! Being in the Panama Canal Zone at the time, it allowed me to
have some contact with the rest of the world - I remember first hearing
and falling in love with Paul Mauriat's "Love Is Blue" and the Beatles'
"Revolution #9" with that radio ("number 9, number 9, number 9, number
9...They are standing still"). I had discovered a whole new universe!
During the same period, I also acquired (via a very painful advance on
my allowance for several weeks) a Sears walkie-talkie (100 mW, channel
14). Oh how I loved that device as well! I used to ride around on my
bicycle trying to find the best place on the base (we were living on
Howard AFB in Panama Canal Zone) for reception. I remember enjoying
hearing even the cacophony of noise that resulted when the skip would
roll in and I'd hear (dozens of?) radio signals fading in and out.
OK, so much for my memory dump... Thanks for listening!
--
% Randy Yates % "...the answer lies within your soul
%% Fuquay-Varina, NC % 'cause no one knows which side
%%% 919-577-9882 % the coin will fall."
%%%% <ya...@ieee.org> % 'Big Wheels', *Out of the Blue*, ELO
http://www.digitalsignallabs.com
John E. Hadstate
"Randy Yates" <ya...@ieee.org> wrote in message
news:m3hcde7...@ieee.org...
> Rick Lyons <R.Lyons@_BOGUS_ieee.org> writes:
>
> During the same period, I also acquired (via a very painful
> advance on
> my allowance for several weeks) a Sears walkie-talkie (100
> mW, channel
> 14). Oh how I loved that device as well! I used to ride
> around on my
> bicycle trying to find the best place on the base (we were
> living on
> Howard AFB in Panama Canal Zone) for reception. I remember
> enjoying
> hearing even the cacophony of noise that resulted when the
> skip would
> roll in and I'd hear (dozens of?) radio signals fading in and
> out.
I'll second that. My brother and I had a pair of those with a
superregenerative detector. If you adjusted the regen
carefully (just before the point where it broke into
oscillation), you could sometimes hear CBers in Louisiana and
Alabama (due, no doubt, to the glories of the Kilowatt Linear).
In retrospect, talking back to them on 100 mw would have been
quite a feat, but that didn't stop us from trying ;-)
Jerry Avins
...
> You can poo-poo this sort of evaluation. But that doesn't make you
> right. Do you have any "proof" that no one can hear the difference?
> Do you even know what the differences are that I was talking about?
What is your opinion of the benefit of using 0-0 gauge gold-plated Litz
wire or flat braid for speaker cables? How about CD demagnetizers? There
are many listeners' claims for theose also.
...
Jerry
--
Engineering is the art of making what you want from things you can get.
¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯
Jerry Avins
...
> now, in *processing* sounds with some nasty non-linearities in the
> process, it very well may be necessary to upsample to 192 kHz or
> higher to do that non-linear processing, and when it is done, LPF to
> 20 kHz and downsample back to 48 kHz. but, except for experimental
> purposes, 192 kHz storage or transimssion is not necessary.
You're wearing an engineer's hat. Now try to think like a marketeer. :-)
Jerry Avins
> rickman <gnu...@gmail.com> writes:
>> [...]
>> When I press the keys on the right, I
>> hear more of a click than a ping (maybe it's the piano)!
>
I can hear all but the top two notes on a piano. My hearing is now down
at 4 KHz 40 dB in one ear and 45 in the other.
> http://en.wikipedia.org/wiki/Piano_key_frequencies
>
> You are free to think what you want about hearing. I choose to side with
> objective measurements verifying again and again over several decades
> the same conclusions rather than a few crackpot audiophools that claim
> they're different without any supporting objective evidence.
>
> And even if one or two actually could hear beyond 20 kHz, they're the
> Robert Wadlow's of the audio world - should we start building houses
> with 10-foot ceilings because 1 out of a billion will be over 8 feet
> tall?
Why not, is you can sell them at a premium? My house sits on a
third-acre lot. Some houses built near here look like they have a
quarter acre footprint. Who needs a McMansion for three people?
Jerry Avins
> Randy Yates wrote:
>
>> The curve on p.20 of
>>
>> http://www.dohc.ie/publications/pdf/hearing.pdf?direct=1
>>
>> indicates that, even for young adults, sound at <20 kHz is
>> inaudible. Based on this, a 44.1 kHz sample rate should be ample.
>
> perceived with ears, but with the full body.
>
> One specific issue are the sudden transitions of
> pitches that real world instruments can create,
> and, possibly, are picked up by different means.
> For example, the sound waves going through the
> skull, brain tissue and then to ears (from inside)
> are "distorted" and, maybe, remodulated, so that
> a person *could* receive "signals" (from real
> instruments), which otherwise would be wiped out
> by sampling low-pass filter, in case of 44.1KHz
> sampling rate.
>
> That's the theory.
> Is this true? I don't know, maybe, maybe not.
> There are people swearing this is the case, others
> say that's nonsense.
> There are people citing Dolby technologies taking
> into account these alternate perception paths, but
> I could not find any real reference.
>
> Nevertheless, one thing is sure (and proven) sound
> waves are not only "eared", they're generally
> perceived by the full body, but the studies I know
> always refer to bass and not to ultrasounds.
I recently had my hearing tested, both through the ear canals and via
bone conduction. The results match to within a few dB, indicating that
my loss of cochlear or nerve, rather than associated with eardrum or
ossicles. I don't think the alternate paths account for much in general.
Jerry Avins
> On May 3, 5:42 pm, Piergiorgio Sartor
> <piergiorgio.sartor.this.should.not.be.u...@nexgo.REMOVETHIS.de>
>>
>> --
>>
>> piergiorgio
>
> I agree that humans cant hear above 20k rate
>
> Its not always about what you hear or percieve with your body.
>
> Its also about how you store data.
>
> here is an simplified analogy.
>
> say you need 44.1k samples per second to hear properly. If the disk
> is corrupted with scrathes
> and 1 samples in his region are lost your sound is distorted or
> lost for that period of time.
> Now if there are 196k samples even if (196/44.1) samples are lost
> there is no difference to what you
> hear.
>
> DVD's come wih high density of data due to this they are highly
> vulnerable to scratches this can be avoided
> with better waveform matching achieved by high sampling rate.
Oh come now! How often do you suppose that one might lose alternate
samples?
dpierce.ca...@gmail.com
Wrong question. That's not the way it works.
Can I PROVE the mnoon is not made of green
cheese, and that NO part of the moon is made
of green cheese? Nope, rather the proof is up to
those who claim it is. Can I PROVE that NO human
can self-levitate? Nope: it's not the job of me or anyone
else to prove that YOUR extraordinary claim is false,
it's YOUR job, YOUR responsibility, if you want to
be taken at all seriously, to prove the extraordinary
claim.
Do YOU have ant proof that one can hear such
differences? No anecdotes, no "I had a dealer friend
who ...," rather verifiable, repeatable, credible proof.
Given the extraordinary amount of science and
engineering that has been spent testing and
describing acoustical psychophysics, for someonem,
such as yourself, to come along with a claim that,
on its face, seems to contradict well over a century
of research, seems to constitute an extraordinary
claim.
Fine, you may be right. But it's not up to the rest
of the worl to prove you wrong, it's up to you to provide
the proof you're right.
That's the way science and technology are
SUPPOSED to work. Unfortunately, these pronciples
and methods are rather inconvenient for much of the
audio world.
To f***ing bad.
dpierce.ca...@gmail.com
Ethan Winer
> This seems to indicate that the tests do show the higher formats to be
> equivalent to CDs.
Yes, which is the main point of the tests.
> Maybe I just don't understand this report, but I see sections where they
> claim that the blind tests showed 100% correct results??!!!
That addresses background noise only, not audio quality. As I read it, they
found only one hi-res disc that had a noise level lower than regular CDs.
And they had to turn up the playback volume way higher than normal to hear
that.
> The question is *not* about ultrasonic content. The question is about
> recording and playback systems using different sample rates.
Actually, the real question is why people sometimes report hearing a change
in audio quality when no change is possible. As with tiny magic stick-on
plastic dots, replacement AC power cords, cryo treatment, and so forth.
That's the core issue underlying ALL of this stuff. Which is what my article
addresses. Did you read my article? Most people - even many audio pros -
have no idea how pervasive comb filtering is.
> You are free to buy any system you want. Why do you care what other
> people use?
This is an excellent question. People who are already convinced their BS
tweaks are worthwhile should buy what they want and I have no objection. But
the majority of people are seeking an honest opinion to avoid wasting money.
I see this as a consumerist issue, so I aim to educate those who really want
to know.
--Ethan
dpierce.ca...@gmail.com
> Its also about how you store data.
>
> here is an simplified analogy.
Yes, simplified to the point of being factually wrong.
> say you need 44.1k samples per second to hear properly.
> If the disk is corrupted with scrathes and 1 samples in his
> region are lost your sound is distorted or lost for that period
> of time.
Wrong. First, you have a pretty robust error correction
scheme built in to the disk. The encoding and decoding
is such that significant amounts of data can be lost
but can be EXACTLY reconstructed on playback with NO
loss. And if the disk is severely scratched to the point where
the error correction algorith fails, interpolation takes place.
One can see thousands of uncorrected errors in the raw
data coming of the disk, and once the error correction
has been applied, the result might be a SMALL handful
(like, oh, 4?) uncorrectable but interpolated errors
> Now if there are 196k samples even if (196/44.1)
> samples are lost there is no difference to what you
> hear.
False. Since you're cramming more data into the same
area, and the physical faults take up the same area
regardless of the data density, more bits, according to
YOUR theory, will be lost on the higher density disk
than on the lower density disk.
That means MORE data is missing, that means the
error correction algorith is subject to higher rates of
non-correctable errors, and so on. Your theory is
bogus if for no other reason than it simply ignores the
facts.
But, in EITHER case, unless the disk is SERIOUSLY
damaged, the data loss in either case is repaired.
> DVD's come wih high density of data due to this
> they are highly vulnerable to scratches this can
> be avoided with better waveform matching achieved
> by high sampling rate.
Sorry, this is nothing but technobabble nonsense.
dpierce.ca...@gmail.com
> On May 3, 8:11 am, Vladimir Vassilevsky <antispam_bo...@hotmail.com>
> wrote:
>
> > rickman wrote:
> > >>Utter nonsense - unless of course you can cite some proper tests.
>
> > > And what do you base this statement on?
>
> > The ultimate reason for the audio systems is making the people happy. If
> > someone is happy because of 192kHz sample rate, and willing to pay for
> > that, then why do you need to proove anything? Heck, if someone orders a
> > 192MHz audio system, it would be my pleasure to do this project.
>
Welcome to the world of high-end audio.
Piergiorgio Sartor
> I recently had my hearing tested, both through the ear canals and via
> bone conduction. The results match to within a few dB, indicating that
> my loss of cochlear or nerve, rather than associated with eardrum or
> ossicles. I don't think the alternate paths account for much in general.
Uhm, but the alternate paths, usually, are no so
specific as per hearing test.
There are several issues here.
One is to distinguish, and it is not easy, between
what we can hear and what "sounds good".
The two things may not be equivalent.
There is a difference between:
1) Attending live a Wagner's concert, in which, for the
record, the orchestra needs some added instruments.
2) Listening the same concert in front of loudspeakers,
which can "only" reproduce up to 20KHz.
3) Having some headphones beaming a pure tone directly
into the head, with someone asking to press a button when
something is heard.
The first is like bathing in a ocean of sound.
The second is like swimming in a pool of sound.
The third is like measuring the water's temperature.
And we, here, are discussing how warm or cold humans
can feel the water.
The problem is how nice or not is the water...
bye,
--
piergiorgio
Robert Adams
<piergiorgio.sartor.this.should.not.be.u...@nexgo.REMOVETHIS.de>
wrote:
I realize that this is a bit off-topic, but;
It seems silly to argue about whether .01% of the population might, on
a good day and with a particular recording, discern the difference
between 44.1K and 192K sample-rates 51% of the time, when the real
reason that recorded sound cannot come close to reproducing live sound
is the nature of 2-speaker stereo itself.
Have you ever wondered why, when you walk into the lobby of a hotel,
that you can immediately tell the difference between live and recorded
music (besides the fact that the singer is off-key) ?
I have my own theory, and it's quite simple. In a live recording, the
radiation pattern of every instrument is unique, with live cymbals in
particular standing out as having a very different radiation pattern
from other instruments. In a stereo playback environment, all
instruments have the same radiation pattern (= the speakers pattern).
The ear discerns this quite easily. If you replace the hotel band with
a recording of the same hotel band, played back in the same location
over PA speakers, you will not have the same "live" impression when
you enter the hotel.
Now I know that if you believe in HRTF's, and your living room in an
anechoic chamber, and you sit glued to one spot, and hrtf curves were
actually gathered from YOUR OWN ears, you should, in theory, be able
to exactly reproduce the pressure waveform at your eardrum that would
have existed in a live performance. But this is so far from the usual,
and practical, case, that it is not worth spending much time on.
Headphones offer a better opportunity for this type of listening, but
again you have the "those are not MY HRTF's" problem.
I don't know a practical solution to this problem, other than to go to
lots of acoustic concerts.
Bob Adams
DigitalSignal
signals with 140 dB SNR? I have never seen one. I guess they are not
using any digital technology, or they are look at very low frequency
signals. Right?
James
www.go-ci.com
emeb
> For people who say supersonic sound can't play a part in a listening
> experience, trying being in a room with a high intensity of supersonic
> energy. Under some conditions (I'm not clear which) you can sense it,
> even though you can't hear it. It actually feels like something loud
> that you can't hear is going on. Its a very odd feeling.
I've noticed this too. Some relatives of mine have an ultrasonic dog
deterrent which looks like a TV remote. Pushing the button on that
thing produces a sensation of intense pressure, even though I can't
hear above about 13kHz.
My guess about what's happening is this: Even though my cochlea isn't
responding to the high frequency, the mechanical linkage of the inner
ear is still transmitting that ultrasonic energy. Perhaps there are
other 'side channel' senses involved - aren't there muscles on the
tympanum that provide a sort of AGC function? Maybe the control loop
that drives them has a wider sensing bandwidth than the nerves that
transmit frequency information to the brain.
Eric
rickman
Thales Communications, makers of military radios. This figure came up
in the context of digital interference with the RF sections. I was
told that the digital noise had to be below -140 dB to prevent
desensitization of the receiver. These units are *very* overbuilt and
far surpass anything commercial I have seen. One spur anywhere in a
huge range, 30 MHz to 500 MHz, IIRC and they are back to the testing
room to add more copper tape or to put more resistors in clock lines,
etc.
Don Pearce
wrote:
A desensitization level is going to be measured in dBm, and -140 is a
very ordinary figure. If external noise interference is present at the
input of a receiver, it will add to the inherent noise already there,
raising it. This is called desensitization. A common interference
requirement is that a desensitization of no more than 1dB is
permitted.
The -140 in this instance is just a receiver input level, and nothing
to do with S/N ratios, dynamic ranges or anything like that.
I would be very surprised if anything military surpassed the spec of a
commercial design. Far too few units are built to justify the number
of trips round the design cycle that are needed to remove every spur
from and maximise performance. That fact that they have to bodge units
with copper tape to make them work rather bears this out.
Certainly when I was designing low noise converters for domestic
satellite systems I achieved a guaranteed total noise figure 0f 0.25dB
at 12GHz, and the entire unit had a works cost price of 11 dollars.
Nothing military has ever come close to that kind of performance or
price.
d
--
Pearce Consulting
http://www.pearce.uk.com
Arny Krueger
in message
news:4f0de839-fa35-4c51...@x35g2000hsb.googlegroups.com
> now, long ago, when DSD was just coming out, i
> participated in something like this that an audio guy
> (now an author) named Bob Katz did. this was before Pro
> Tools HD and they were using some expensive system from a
> company called Sonic Solutions that could do 192 kHz.
> they recorded at 192 kHz some test sounds (including some
> high frequency percussive sounds like castenets, cabasas,
> cymbals) along with synthesized bandlimited (to 96 kHz)
> tones of all sorts of waveshapes at a variety of
> frequencies from below 10 kHz up to 90 kHz. i don't
> remember all of the test sounds, but they made sure that
> most, if not all, had content well above 30 kHz. the
> subject listened to those sounds in two different forms,
> but both with a 192 kHz playback rate. one form was the
> raw recorded sound, the other was processed through a
> phase-linear FIR filter with a lot of taps (i thought it
> was around 300 something taps, it was not a real-time FIR
> filter but proceessed one sound file into another) that
> was flat to within 0.01 dB up to 20 kHz, had a smooth
> transition band from 20 to 22 kHz, and then was down by
> more than 130 dB for 22 kHz to 96 kHz. we padded the
> beginning an end of the original sound file with zeros,
> of half the length of the FIR on both ends and the
> non-real-time FIR was not "causal" and had a delay of 0
> (so it started responding before the first non-zero
> samples) and the filtered sound file was lined up in time
> with the original.
> now this isn't what Bob Katz did, but it is what i wished
> he did: called the original sound "A", and the LPFed
> sound "B". so sometimes the subject hears AA, sometimes
> AB, sometimes BA, and sometimes BB (all four permutations
> exist in equal quantity) and for each pair, the subject
> simply has to say if they think the sounds are the same
> or different. every time a subject says that AA or BB
> sound different, we count that as a false positive and
> subtract that count from the number of times the subject
> says that AB or BA sound different. likewise for false
> negatives.
That is called same/difference testing. It is an older form of testing,
valid enough as far as it goes. The most important question in a test is not
the order of presentation of variables (within reason) but rather whether
the listener can determine the identity of the so-called unknowns by any
means other than listening. Obviously, if one can determine the the identity
of the unknows by any means other than listening, the unknowns are not
unknown but known, and there is no test.
> now Bob didn't do that, but he did something like it (i
> think it was ABX) and there was no statistically measured
> difference.
At this point 100's if not 1,000s or perhaps even 10,000s of people have
done this kind of test, and they obtained the same results as Katz. Please
see www.pcabx.com.
> people could simply not reliably tell if the
> stuff above 20 kHz was removed or not. they could not
> tell at all. no one could.
Welcome to the well-known limitations of the human ear.
> now, if they cannot tell if the content existed above 22
> kHz or not, is there a need to have it there in storage
> or in transmission?
No.
> if there is no need to have it
> there, and it is removed, what does the sampling theorem
> tell us regarding sufficient sampling rate?
It tells us that a well-implemented 44 KHz sample rate might even be
overkill.
>> I'm not trying to "prove" anything. I am presenting
>> information which you can consider and believe or can
>> ignore. But you can't say my statements are false unless
>> you have some information to "prove" they are.
Requiring absolute proof can be a sort of excluded-middle argument. IOW
there are a great many things that have not been absolutely proven false,
that are generally and correctly treated as being false. In fact, many many
negative hypotheses, such as the idea that anything in particular is false,
are difficult or impossible to prove.
Just because I can't absolutely prove that pigs can't fly is not a good
reason to start an airline based on flying pigs. ;-)
>> Human hearing is not a microphone connected to an amplifier.
Often human hearing is far less sensitive than a microphone connected to an
amplifier. So what?
>> It is a very complex process which even includes the
>> brain and we certainly don't understand it completely.
We don't have to understand everything completely in order to come up with
effective working hypothesis about them.
> but we can't hear anything above 20 kHz.
Actually, often many of us can hear something above 20 KHz if it is an
isolated tone and also loud enough. The question at hand is about something
very, very different - whether or not we can hear the elimination of
information above 20 KHz from music.
> even with
> percussive sounds with sharp attacks. and if we cannot
> hear anything above 20 kHz, then
> 40.0001 kHz sampling rate can store all of the
> information we need. for practical reconstruction
> purposes, 44.1 and 48 kHz are sufficient.
That turns out to be a hypothesis whose contradiction is difficult or
impossible to prove.
> now, in *processing* sounds with some nasty
> non-linearities in the process, it very well may be
> necessary to upsample to 192 kHz or higher to do that
> non-linear processing, and when it is done, LPF to 20 kHz
> and downsample back to 48 kHz. but, except for
> experimental purposes, 192 kHz storage or transimssion is
> not necessary.
Agreed.
Arny Krueger
news:96f46e02-0219-4e88...@a70g2000hsh.googlegroups.com
> I never said that *anyone* can hear sounds above 20 kHz.
But I'm willing to agree that some people can do this, even in a blind test.
However, its not the question at hand.
> I said that there are people who can hear the difference
> between systems that use higher sample rates than 48 kHz.
I'm willing to agree that this can happen, but the explanations always seem
to have this annoying tendency to not be the same as the question at hand.
Just because a system uses a sample rate >= 48 kHz does not mean that it is
free of audible flaws.
The question at hand is whether or not introducing an ideal low pass filter
is audible when listening to music, and when that low pass filter has
negligable effects below 20 kHz.
For example, a - 6 dB/octave filter with a - 3dB point at 20 kHz also causes
1 dB loss at 10 KHz, which some listeners can detect.
> The real issue is about the electronics and how they are
> measured *compared* to human hearing. Distortion is the
> real issue. Filters introduce distortion and some people
> can hear the distortion produced by converters running at 48 kHz.
There are basically only two types of distortion - linear distortion and
nonlinear distortion. Since you have not restricted your comment to
nonlinear distortion, there is a possiblity that you are talking about
nonlinear distortion, which is again not the question at hand.
> I suppose it is possible that my friend was mislead (and
> therefor myself).
Actually, it is for all known practical purposes, a certainty.
> I have no first hand knowledge of the
> sound tests.
Many of us do.
Please see www.pcabx.com .
> But he is not a person to believe in rubbish.
Well, up until this question... ;-)
> In fact, he is a person who is very intellectually critical.
Apparently, not critical and well-informed enough.
> This is one of the few times he
> could not dismiss claims that run counter to the science.
That's his problem.
robert bristow-johnson
> "robert bristow-johnson" <r...@audioimagination.com> wrote
> in messagenews:4f0de839-fa35-4c51...@x35g2000hsb.googlegroups.com
>
> > now this isn't what Bob Katz did, but it is what i wished
> > he did: called the original sound "A", and the LPFed
> > sound "B". so sometimes the subject hears AA, sometimes
> > AB, sometimes BA, and sometimes BB (all four permutations
> > exist in equal quantity) and for each pair, the subject
> > simply has to say if they think the sounds are the same
> > or different. every time a subject says that AA or BB
> > sound different, we count that as a false positive and
> > subtract that count from the number of times the subject
> > says that AB or BA sound different. likewise for false
> > negatives.
>
> That is called same/difference testing.
thanks, i'll use that term in the future.
> It is an older form of testing,
> valid enough as far as it goes. The most important question in a test is not
> the order of presentation of variables (within reason) but rather whether
> the listener can determine the identity of the so-called unknowns by any
> means other than listening.
that's what blind testing is about.
>
> > now Bob didn't do that, but he did something like it (i
> > think it was ABX) and there was no statistically measured
> > difference.
>
> At this point 100's if not 1,000s or perhaps even 10,000s of people have
> done this kind of test, and they obtained the same results as Katz. Please
> see www.pcabx.com.
i have never completely accepted that ABX testing is better than "same/
difference testing" (what i used to call "AB testing") for the
question we are trying to answer here.
ABX is for answering the question: which is better? or which is
closer? does synthesizer A sound more like a real piano or does
synthesizer B sound more like the real piano?
even though i am politically very liberal, as an engineer, i am *very*
conservative. say the issue is Monster Cable. the claim is that
Monster Cable sounds better than 10 gauge lamp cable. now, being
conservative, i am willing to grant Monster Cable the benefit of doubt
in that if people can actually hear a *difference*, i'm willing to let
their claim that they sound better go without challenge. but if
people *cannot* hear a difference, then i am not willing to accept,
even tentatively, that Monster Cable sounds better. for it to sound
better, it has to at the very least, sound different. since "better"
and "worse" is a subjective thing (like clean vinyl vs. CD), i am too
conservative to take a position regarding which is "better" if they're
different. but i am not too conservative to say "bunk" about
something being experienced as "better" if it is not experienced as
different.
"same/difference testing" does not cloud the issue with an ancillary
question of "which one of these two sound closer to a third?" it only
asks the question "do these sound discernably different or do they
not?" and with same/difference testing, it is so easy to unbias the
measure so we can even invite a Monster Cable salesperson to the test
and let them know that if they guess with a bias that these sounds
*do* sound different (which would support their case, but not fairly),
we will know it (because we subtract the false positives from the true
positives). i think for this kind of question, the Monster Cable
sounds better or ultra-high sampling rates sound better or whatever,
same/difference testing speaks to that question directly whereas ABX
is not as direct.
r b-j
geoff
> If it really is a waste of time and money to use 192 kHz ADC and DAC,
> why do you think they would do it? Don't you think the people
> designing DVD equipment understand the economics of consumer
> products?
>
> Try to think about it and see if you can come up with a couple of
> reasons yourself. I'll be interested in hearing what you think.
There are 3 reasons why people design and manufacture 192KHz equipment:
1 - They imagine it makes a difference.
2 - The technology is available, so why not
3 - Everybody already has 44k1/48K gear, so what would we sell them
otherwise....
geoff
Arny Krueger
> On May 3, 3:35 pm, Oli Charlesworth
> <ca...@olifilth.co.uk> wrote:
>> On May 3, 5:26 pm, dpl...@radagast.org (Dave Platt)
>> wrote:
>>
>>
>>
>>> There may be _some_ justification for it, as it
>>> eliminates the need to place the knee of the
>>> anti-aliasing filter anywhere near the range of
>>> frequencies that one _can_ hear. One of the criticisms
>>> made against CD is that the sharp filtering which must
>>> be done at around 20 kHz can cause artifacts which may
>>> be audible to some listeners, either due to
>>> "pre-ringing" (with a symmetric FIR low-lass filter) or
>>> a frequency- dependent delay and "smearing" of
>>> transients (with an IIR filter).
>>
>>> These effects can be moved up to higher frequencies,
>>> and prevented from having effects in the human hearing
>>> passband, by increasing the sampling rate.
>>
>> Indeed. But this problem is largely solved by
>> oversampling at the DAC, without requiring an increased
>> sample rate at the storage-medium level.
Right, the data that is stored might be for example at 44,100 Hz, but the
oversampling process involves temporarily using a sampling rate of for
example 5 times that or about 220,500 Hz.
> And how exactly do you over sample at the DAC without a
> filter?
There is still a filter, but its at a frequency related to the oversampling,
not the sampling.
Arny Krueger
> i have never completely accepted that ABX testing is
> better than "same/difference testing" (what i used to call "AB testing")
> for the question we are trying to answer here.
ABX is based on same/difference testing. ABX facilitates same/difference
testing.
> ABX is for answering the question: which is better?
Not really. It is for answering the question: are they the same or
different?
> or which is closer? does synthesizer A sound more like a real piano
> or does synthesizer B sound more like the real piano?
The ABC/hr test does a better job then ABX at doing that sort of thing.
> even though i am politically very liberal, as an
> engineer, i am *very* conservative.
No problem.
> say the issue is Monster Cable. the claim
> is that Monster Cable sounds better than 10 gauge lamp cable.
The more reasonable claim would be that Monster Cable sounds better than 13
gauge lamp cable because much of the better Monster speaker cables are the
equivalent of 13 gauge stranded wire.
> now, being conservative, i am willing to grant Monster Cable the
> benefit of doubt in that if people can actually hear a *difference*, i'm
> willing to let their claim that they sound better go without challenge.
I wouldn't call that being conservative. My flavor of conservatism would
oblige me to test for better sound, even if I found that they sounded
different.
> but if people *cannot* hear a difference, then i am not willing
> to accept, > even tentatively, that Monster Cable sounds better.
That's just common sense. Sounding better implies sounding different. No
difference, no better sound. Logic.
> for it to sound better, it has to at the very least, sound different.
Agreed.
> since "better"
> and "worse" is a subjective thing (like clean vinyl vs. CD),
Another view is that any uncontrollable change is a degradation. IOW louder
or softer or the intentional application of a tonal shift is OK, but if an
inherent and irreducable result of some operation, then that change is
undesirable.
Since it is impossible to avoid audible changes during recording and playing
back using the LP format, the difference has to be undesirable.
Arny Krueger
news:M5CdnfvR7O51q4PV...@giganews.com
That is pretty well it. As time marches on, 192/24 converters with > 115 dB
dynamic range are becoming jelly bean (highly inexpensive) chips. A number
of years back us quality bugs smiled and ponied up about $800 for Lynx L33
cards, but now you can get pretty much the same converters and performance
in < $200 eMu cards.
rickman
I am not an RF guy and the figure I actually remember was 150 dB. I
hedged it a bit as 150 sounded rather extreme to me. I dunno if 150
is anything outside of ordinary or not.
When you say that it costs too much to go around the design cycle many
times, you clearly don't know much about the military procurement
process. On the last radio that they built while I was there, they
were up to rev 14 of the board that had nothing but the UI controller
and external interfaces. You need to remember that often the
development process is cost plus and the customer is *asking* for
tough specs. It is only when they can't be delivered that they back
off.
What is really funny is that you are getting wrapped around the axle
about my use of this figure when that was really just an aside to an
aside of my original point. Funny how these discussions get so far
off topic.
Did you read my post which used the original 140 dB figure?
Don Pearce
wrote:
OK 150 - different bandwidth, then. The basic thermal noise at the
front end of a receiver is -174dBm + 10 log(bandwidth) + noise figure.
A noise or interference level that desensitizes that by 1dB will be
about 10dB lower than that level. You need to do the know the
bandwidth and noise figure to know the significance.
What is significant is that this receiver has a desensitization limit
for interference that it throws at itself. I have never, ever come
across that before - it is always an external spec; you design the
internals so it doesn't interfere.
>When you say that it costs too much to go around the design cycle many
>times, you clearly don't know much about the military procurement
>process. On the last radio that they built while I was there, they
>were up to rev 14 of the board that had nothing but the UI controller
>and external interfaces. You need to remember that often the
>development process is cost plus and the customer is *asking* for
>tough specs. It is only when they can't be delivered that they back
>off.
>
Iteration 14 and the company was still in business? Anyone can afford
to do it again if they can't do it right. Was this a "cost-plus"
contract? Three or perhaps four was more the number I had in mind.
>What is really funny is that you are getting wrapped around the axle
>about my use of this figure when that was really just an aside to an
>aside of my original point. Funny how these discussions get so far
>off topic.
>
If you like. It was clearly a number you threw in because it sounded
impressively big, even though you hadn't a clue what it meant.
>Did you read my post which used the original 140 dB figure?
Yes. It was nonsense.
rajesh
Thanks ! Your facts are proving my point.
Repeating samples is the most simplest form of error correcting codes.
All your error correcting codes and interpolation techniques become
196/44.1 folds
more robust on 196 kHz signal compared 44.1 kHz signal.
You just have to accept this point of view although it may not justify
for going 196 kHz.
" remembering and quoting facts is no big deal, you have to learn to
analyze them"
rajesh
Try writing 44.1 khz signal on that high density disc(greed to store
more music)....small scratch and
and the disc is busted.
rajesh
Remember the shannon's theorem which places a trade off between error
correcting codes and bandwidth.
Mr.T
"Green Xenon [Radium]" <gluc...@excite.com> wrote in message
news:481becfe$0$5141$4c36...@roadrunner.com...
> Why does DVD-Audio use 192 kHz sample rate? What's the advantage over
> 44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate?
This is the same guy who wanted a 3GHz sample rate in an earlier post!!!!
> On average, what is the minimum sample rate for a guy in his early to
> mid 20s who likes treble?
44.1 ks/s.
> I agree there are a small percentage of humans who can hear above 20
> kHz. However, DVD-audio uses a sample-rate of 192 kHz which allows a
> maximum frequency of 96 kHz. There is no known case of any human being
> able to hear sounds nearly as high as 96 kHz. I can agree with 48 kHz
> sample rate and even 96 kHz sample-rate [maybe], but 192 kHz is just
stupid.
>
> So whats the justification fur using 192 kHz? If you ask me, its just a
> total waste of bandwidth and energy. Any proof to the contrary?
The advertising sounds better.
> Please correct me if I'm wrong but AFAIK, its a waste of time, money,
> energy to move to 192 kHz.
So why did you want 3GHz for audio then!!!!!!!!!!!!!!
Please explain your sudden change of heart.
(and yes I know he's just a troll)
MrT.
Mr.T
> If it really is a waste of time and money to use 192 kHz ADC and DAC,
> why do you think they would do it? Don't you think the people
> designing DVD equipment understand the economics of consumer
> products?
>
> Try to think about it and see if you can come up with a couple of
> reasons yourself. I'll be interested in hearing what you think.
Because it costs them no more and the advertising sounds better to the
uninformed.
What did you come up with?
MrT.
Randy Yates
> [...]
> Remember the shannon's theorem which places a trade off between error
> correcting codes and bandwidth.
No, Shannon's thereom places an upper limit on the rate at which
one can reliably communicate in a white Gaussian noise channel
based on bandwidth, signal power, and noise power.
--Randy
@article{shannon,
title = "Communication in the Presence of Noise",
author = "Claude E. Shannon",
journal = "Proceedings of the Institute of Radio Engineers",
year = "1949",
volume = "37",
pages = "10-21"}
--
% Randy Yates % "Bird, on the wing,
%% Fuquay-Varina, NC % goes floating by
%%% 919-577-9882 % but there's a teardrop in his eye..."
%%%% <ya...@ieee.org> % 'One Summer Dream', *Face The Music*, ELO
http://www.digitalsignallabs.com
Oli Charlesworth
No, they don't. If the same FEC techniques are used on both discs,
then for the same proportion of raw errors (read errors), the number
of uncorrectable errors will be the same. However, as dpierce already
pointed out, given the same physical damage to both discs, the high-
density disc will experience a proportionally higher density of raw
errors, therefore the number of uncorrected errors will be higher.
The same logic applies to interpolation techniques. Your
"explanation" could only work if the high-density player was designed
to interpolate over errors and then downsample to 44.1.
--
Oli
dpierce.ca...@gmail.com
> On May 4, 8:10 pm, dpierce.cartchunk....@gmail.com wrote:
> > Sorry, this is nothing but technobabble nonsense.
>
> Thanks ! Your facts are proving my point.
Yes, and you continue below to spew pure nonsense.
> Repeating samples is the most simplest form of
> error correcting codes.
Repeated samples ARE NOT "error correcting codes."
And increasing the sample rate DOES NOT "repeat
samples.
In both cases, you deomnstrate your complete lack of
understanding of the principles involved..
> All your error correcting codes and interpolation
> techniques become 196/44.1 folds
> more robust on 196 kHz signal compared 44.1 kHz signal.
No, they do not.
> " remembering and quoting facts is no big deal,
> you have to learn to analyze them"
And when people like deomstrate a complete lack
of understanding of the facts, yous substitute just
making sh*t up. That's better how?
dpierce.ca...@gmail.com
> Remember the shannon's theorem which places a
> trade off between error correcting codes and bandwidth.
Again, pure nonsense. Shannon's theorem
never discusses error correcting codes AT ALL.
rajesh
hi pierce ,
BTW from which school did u learn DSP?
Robert Lacoste
news: 481becfe$0$5141$4c36...@roadrunner.com...
> Hi:
>
> Why does DVD-Audio use 192 kHz sample rate? What's the advantage over 44.1
> kHz? Humans can't hear the full range of a 192 kHz sample rate?
I think that the answer is aliasing avoidance. Take it this way :
- The audio band pass is limited to 16KHz, say 20KHz to get some extra
marging for the most perfect ears on earth.
- As far as I know ANY audio digitization circuit uses a low pass filter at
around 20KHz, so even a 192Ksps ADC or DAC will be band pass limited to
20KHz signals, as there is absolutely no need to manage audio signals with a
higher frequency.
- If you use 44Ksps then you must insure that there is no power above
44/2=22KHz thanks to M. Nyquist, so your low pass filter must have a very
sharp transition. As the filter will never be perfect you will get aliases.
For example even if you use a 12th order filter (already difficult and
expensive to build) then the attenuation will be "only" 72dB/octave, meaning
that a 16KHz low pass filter will have an attenuation of only 50dB or so at
22KHz. And 50dB is not enough for good listeners as a -50dBc "noise" is
clearly audible.
- However if you use a 192Kbps sampling rate then the required performances
on the low pass filter are drastically relaxed. This filter can keep a
corner frequency at 16 or 20KHz, but even a 6th order filter will provide a
at 86dB attenuation at 192/2=96KHz...
And as a 192Ksps sampling rate is far cheaper to build than a very very good
low pass filter... That's the beauty of oversampling...
Does it make sense ?
Cheers,
Robert Lacoste
www.alciom.com
The mixed signal experts
Don Pearce
Rajesh, please take some advice and don't do this. He is going to rip
you to pieces and embarrass you badly.
Don Pearce
Not a lot. As far as I'm aware there are NO ADCs that sample at the
data rate of the output signal. For example the 44.1ksps ADC in my PC
samples at 2.8224MHz. When you sample at that rate it is trivially
easy to make a gently sloping analogue lowpass filter that guarantees
a lack of alias products. All further filtering and decimation is done
digitally, where it is easy. THAT is what oversampling is all about,
not using a 192ksps sampling rate.
rajesh
Here is the coz for mis interpretation of my opinions.
I am not comparing two different discs of two different capacities at
all.
I am comparing, given the same high capacity disc, which sampling rate
can cope better with errors.
And the answer is the one with which is oversampled.
say if u are comparing 44.1 kHz CD and a 196kHz DVD then they both
cope with errors equally.there is no doubt about this.
But then in the later case you are not comparing apples to apples.
Oli Charlesworth
for-email> wrote:
> "Green Xenon [Radium]" <gluceg...@excite.com> a écrit dans le message denews: 481becfe$0$5141$4c368__BEGIN_MASK_n#9g02mG7!__...__END_MASK_i?a63jfAD$z...@roadrunner.com...
> - However if you use a 192Kbps sampling rate then the required performances
> on the low pass filter are drastically relaxed. This filter can keep a
> corner frequency at 16 or 20KHz, but even a 6th order filter will provide a
> at 86dB attenuation at 192/2=96KHz...
>
> And as a 192Ksps sampling rate is far cheaper to build than a very very good
> low pass filter... That's the beauty of oversampling...
Oversampled conversion does not require one to *store* information at
the oversampled rate.
--
Oli
Don Pearce
<getra...@gmail.com> wrote:
>On May 5, 4:21 pm, nos...@nospam.com (Don Pearce) wrote:
>> On Mon, 5 May 2008 04:15:59 -0700 (PDT), rajesh
>>
>> <getra...@gmail.com> wrote:
>> >On May 5, 3:46 pm, dpierce.cartchunk....@gmail.com wrote:
>> >> On May 5, 4:05 am, rajesh <getrajes...@gmail.com> wrote:
>>
>> >> > Remember the shannon's theorem which places a
>> >> > trade off between error correcting codes and bandwidth.
>>
>> >> Again, pure nonsense. Shannon's theorem
>> >> never discusses error correcting codes AT ALL.
>>
>> >hi pierce ,
>>
>> >BTW from which school did u learn DSP?
>>
>> Rajesh, please take some advice and don't do this. He is going to rip
>> you to pieces and embarrass you badly.
>>
>> d
>>
>> --
>> Pearce Consultinghttp://www.pearce.uk.com
>
>Here is the coz for mis interpretation of my opinions.
>
>I am not comparing two different discs of two different capacities at
>all.
>
>I am comparing, given the same high capacity disc, which sampling rate
>can cope better with errors.
>
>And the answer is the one with which is oversampled.
>
Do you even know what oversampling is? You won't find it on any disc -
it is a function of the front end of an ADC.
And of course they are ALL oversampled, no exceptions, ever.
>
>
>say if u are comparing 44.1 kHz CD and a 196kHz DVD then they both
>cope with errors equally.there is no doubt about this.
>
>But then in the later case you are not comparing apples to apples.
>
The way they cope with errors is a function of the error
detection/correction schemes they use.
rajesh
> On Mon, 5 May 2008 04:36:01 -0700 (PDT), rajesh
>
>
>
> <getrajes...@gmail.com> wrote:
> >On May 5, 4:21 pm, nos...@nospam.com (Don Pearce) wrote:
> >> On Mon, 5 May 2008 04:15:59 -0700 (PDT), rajesh
>
Assume for a moment there are no error correction codes.
Now which one is better.
error correction codes dont care about the signal's sampling rate or
what ever.
rajesh
> On Mon, 5 May 2008 04:36:01 -0700 (PDT), rajesh
>
>
>
> <getrajes...@gmail.com> wrote:
> >On May 5, 4:21 pm, nos...@nospam.com (Don Pearce) wrote:
> >> On Mon, 5 May 2008 04:15:59 -0700 (PDT), rajesh
>
Mr Pearce please view my profile and read some of my posts.
Don Pearce
There will be no competition. They will both be unusable crap. Error
correction is a vital and integral part of the storage process.
Don Pearce
I have no idea what you mean by your profile. And those posts of yours
that I have so far read make it clear that you don't even know the
meaning of the word oversampling. That is pretty fundamental to an
understanding of DSP, so no, my money is not on you in this race.
rajesh
> On Mon, 5 May 2008 04:45:57 -0700 (PDT), rajesh
>
>
>
I dont think you are tryin to comprehend my question
rather giving unruly answers
Oli Charlesworth
It is difficult to determine what your current question is, as your
train of replies has been very muddled. However, would it be fair to
say that your current question is "which would be better given no
error-correcting: a 192kHz disc or a 44.1kHz disc?"
Aside from the fact that has already been pointed out (they would both
be unusable), there is no reason to suggest that the 192kHz disc would
be any better.
--
Oli
Randy Yates
> [...]
> Repeated samples ARE NOT "error correcting codes."
Yes they are, Dick. They are commonly called "repetition codes." See
for example one of the following:
@book{berlekamp,
title = "{Algebraic Coding Theory}",
author = "{Elwyn R. Berlekamp}",
publisher = "Aegean Park Press",
edition = "revised 1984 edition",
year = "1984"}
@book{wicker,
title = "Error Control Systems for Digital Communication and Storage",
author = "Stephen B. Wicker",
publisher = "Prentice Hall",
year = "1995"}
> [...]
> And increasing the sample rate DOES NOT "repeat
> samples.
I agree with this, so the point above is probably moot.
--
% Randy Yates % "I met someone who looks alot like you,
%% Fuquay-Varina, NC % she does the things you do,
%%% 919-577-9882 % but she is an IBM."
%%%% <ya...@ieee.org> % 'Yours Truly, 2095', *Time*, ELO
http://www.digitalsignallabs.com
rajesh
in one case the data written is that of 192khz signal and other case
it is 44.1 khz. please clarify this.
So if you apply same error correcting code on both cases , you will
have a reason to say that 192khz is better.
Don Pearce
wrote:
>dpierce.ca...@gmail.com writes:
>> [...]
>> Repeated samples ARE NOT "error correcting codes."
>
>Yes they are, Dick. They are commonly called "repetition codes." See
>for example one of the following:
>
>@book{berlekamp,
> title = "{Algebraic Coding Theory}",
> author = "{Elwyn R. Berlekamp}",
> publisher = "Aegean Park Press",
> edition = "revised 1984 edition",
> year = "1984"}
>@book{wicker,
> title = "Error Control Systems for Digital Communication and Storage",
> author = "Stephen B. Wicker",
> publisher = "Prentice Hall",
> year = "1995"}
>
>> [...]
>> And increasing the sample rate DOES NOT "repeat
>> samples.
>
>I agree with this, so the point above is probably moot.
But repeating a sample cannot of itself correct errors. What you end
up with is two samples of which one, neither or both may be wrong. Of
course you can use a repeated sample as somewhere to go and get the
correct data if the original sample has already been detected as
incorrect. Still a bit of a leap of faith, though.
The best error correction method is powerfully dependent on the medium
and they way it introduces the errors - single bit errors vs large
block errors for example. For a CD it is all quite easy, as there is
no low latency pressure which you would have in, for example, a VOIP
stream.
Oli Charlesworth
> On May 5, 5:00 pm, Oli Charlesworth <ca...@olifilth.co.uk> wrote:
>
> > On May 5, 12:54 pm, rajesh <getrajes...@gmail.com> wrote:
>
> > > On May 5, 4:49 pm, nos...@nospam.com (Don Pearce) wrote:
>
> > > I dont think you are tryin to comprehend my question
> > > rather giving unruly answers
>
> > It is difficult to determine what your current question is, as your
> > train of replies has been very muddled. However, would it be fair to
> > say that your current question is "which would be better given no
> > error-correcting: a 192kHz disc or a 44.1kHz disc?"
>
> > Aside from the fact that has already been pointed out (they would both
> > be unusable), there is no reason to suggest that the 192kHz disc would
> > be any better.
>
> in one case the data written is that of 192khz signal and other case
> it is 44.1 khz. please clarify this.
In that case it is you who should clarify. We've all been talking
about a comparison between a high-density disc (e.g. a DVD) and a
standard CD. What are you talking about?
> So if you apply same error correcting code on both cases , you will
> have a reason to say that 192khz is better.
Now you're just going round in circles. Do you want to talk about ECC
or not?
--
Oli
rajesh
I dont mean literal repetition.Take for example a sine wave of 10khz
and sample it with 1 mhz. One does think of
samples getting repeated are atleast they are close.
rajesh
I havent read all the posts. But the topic of this thread mentions
only two different sampling rates and it
DOESNT mention any thing about discs
rajesh
"Why does DVD-Audio use 192 kHz sample rate? What's the advantage over
44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate? "
He means to say that why 44.1khz is not used on DVD's?
Thats exactly what i answered
Randy Yates
I didn't see that the number of repeated samples was one from Dick's
post (or the previous poster). But yes, just doing a single repeat
(transmitting two samples for every one) does not buy you anything.
--
% Randy Yates % "Though you ride on the wheels of tomorrow,
%% Fuquay-Varina, NC % you still wander the fields of your
%%% 919-577-9882 % sorrow."
%%%% <ya...@ieee.org> % '21st Century Man', *Time*, ELO
http://www.digitalsignallabs.com
Don Pearce
Utterly wrong. All the samples are unique. Please go and do some
reading on DSP before you continue.