> 
> Please note that many modern function generators are indeed
> software "driven". I.e. a uP or a DSP generates the waveform
> which is then DA-converted and filtered.
> 
> 

I know.
But they are more expensive and for the test I was thinking about a full 
analog one would be enough.

-- 
Laurent ARTAUD (lau### [at] freefr)

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> Yeah, I have software and hardware to play 24/96 from my PC, it's just I
> doubt my speakers could keep up [just looking], apparently they are
> down -3dB at 22kHz, so who knows what they'll be down to at 48kHz.
> 

Well, if you have the software, try this:
- create a sin waveform at 20KHz and save it in a 16/44 format
- create a sin waveform at 20KHz and save it in a 24/96 format
- at 24/96, import the two audio files and compare them, the difference 
should be obvious (if not, try it with 22KHz waveforms).
- if you want to have fun, sub the 24/96 by the 16/44 and play the 
result (at 24/96). Depending of your ears and your hardware, you may be 
able to hear the difference.

Regards,

-- 
Laurent ARTAUD (lau### [at] freefr)

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laurent.artaud[AT]free.fr" <"laurent.artaud[AT]free.fr wrote:
>> Yeah, I have software and hardware to play 24/96 from my PC, it's
>> just I doubt my speakers could keep up [just looking], apparently
>> they are down -3dB at 22kHz, so who knows what they'll be down to at
>> 48kHz.
>>
>
> Well, if you have the software, try this:
> - create a sin waveform at 20KHz and save it in a 16/44 format
> - create a sin waveform at 20KHz and save it in a 24/96 format
> - at 24/96, import the two audio files and compare them, the
> difference should be obvious (if not, try it with 22KHz waveforms).

No.  There is no difference in the waveforms.  See my posts in
povray.binaries.misc.  Or download Adobe Audition demo version to try out
for yourself.

[1] This is a 20kHz sine wave saved and then loaded in 44.1 / 16-bit format
[2] This is file [1] converted to 96 / 32-bit format, then saved and loaded
[3] This is a 20kHz sine wave generated in 96/32 format, saved and loaded
[4] This is the difference between [2] and [3], computed in 32-bit format.

Please note that in [4] the scale goes from -32000 to 32000, so we are
seeing an error of between 1 and 2 per 32000, ie -84dB.  Theoretically there
should be no error, but I had choose a quality setting when I upsampled in
stage 2, so I'm sure that effects the result.

> - if you want to have fun, sub the 24/96 by the 16/44 and play the
> result (at 24/96). Depending of your ears and your hardware, you may
> be able to hear the difference.

I can't hear the -84dB signal on my machine at work, I'll try it at home but
I'd imagine I would need to have my amplifier up *far* higher than I do
normally.

Post a reply to this message

> 
> No.  There is no difference in the waveforms.  See my posts in
> povray.binaries.misc.  Or download Adobe Audition demo version to try out
> for yourself.

Silly me! Of course there is no difference: the filters keep only the 
fundamental, which is a sin at 20KHz! It's a triangular waveform, not a 
sin waveform you have to use.
ARGH!
I'm starting to loose myself!
I can't remember how the test was supposed to be done!
Was it 20KHz or 22KHz? Was it a triangular or a square waveform?

(I can't test it myself: I use Linux and have no sound editing softwares 
installed (I'll have to find one to try, but I will not be able to do 
that before this WE...))
(I can't find p.b.m...)

> 
> Please note that in [4] the scale goes from -32000 to 32000, so we are
> seeing an error of between 1 and 2 per 32000, ie -84dB.  Theoretically there
> should be no error, but I had choose a quality setting when I upsampled in
> stage 2, so I'm sure that effects the result.

Well, In my opinion, this error is just a rounding error during the 
conversions... It is not the one I was talking about.

> 
> I can't hear the -84dB signal on my machine at work, I'll try it at home but
> I'd imagine I would need to have my amplifier up *far* higher than I do
> normally.
> 

Well, if you can hear this signal, you have a really really really 
hi-end audio system. Most of the signal/noise ratios are around 70-80dB, 
so you might hear the amplifier's noise better that the signal.

Regards,

-- 
Laurent ARTAUD (lau### [at] freefr)

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laurent.artaud[AT]free.fr" <"laurent.artaud[AT]free.fr wrote:
>> No.  There is no difference in the waveforms.  See my posts in
>> povray.binaries.misc.  Or download Adobe Audition demo version to
>> try out for yourself.
>
> Silly me! Of course there is no difference: the filters keep only the
> fundamental, which is a sin at 20KHz! It's a triangular waveform, not
> a sin waveform you have to use.
> ARGH!
> I'm starting to loose myself!
> I can't remember how the test was supposed to be done!
> Was it 20KHz or 22KHz? Was it a triangular or a square waveform?

Assuming you filter at 48kHz for a 96kHz sampling rate, a 20kHz square wave
will appear as a 20kHz sine wave because the next harmonic in a square wave
is at 3 times the fundamental.  Can't remember what it is for a triangle
wave though.  Will test out this evening listening to them to see if I can
tell any difference :-)

>> Please note that in [4] the scale goes from -32000 to 32000, so we
>> are seeing an error of between 1 and 2 per 32000, ie -84dB.
>> Theoretically there should be no error, but I had choose a quality
>> setting when I upsampled in stage 2, so I'm sure that effects the
>> result.
>
> Well, In my opinion, this error is just a rounding error during the
> conversions... It is not the one I was talking about.

Yes, I think it is rounding error too.

>> I can't hear the -84dB signal on my machine at work, I'll try it at
>> home but I'd imagine I would need to have my amplifier up *far*
>> higher than I do normally.
>>
>
> Well, if you can hear this signal, you have a really really really
> hi-end audio system. Most of the signal/noise ratios are around
> 70-80dB, so you might hear the amplifier's noise better that the
> signal.


claimed specs:

Frequency response: 10Hz - 80kHz -3dB
SNR: 98dB

Unfortunately I don't think my speakers can keep up with that! (claimed
40-22kHz -3dB)

One thing that I do notice is that if I turn my amp up to full while
connected to my 2496 soundcard I hear no hiss or noise at all.  Switch it to
my SoundBlaster Live and wham! there is so much hiss, even when all outputs
are muted in windows ;-)

Post a reply to this message

> 
> One thing that I do notice is that if I turn my amp up to full while
> connected to my 2496 soundcard I hear no hiss or noise at all.  Switch it to
> my SoundBlaster Live and wham! there is so much hiss, even when all outputs
> are muted in windows ;-)
> 

By the way, which 2496 sound card do you have?

Regards,

-- 
Laurent ARTAUD (lau### [at] freefr)

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in news:404e3ba9$1@news.povray.org Severi Salminen wrote:

> I don't think there are many printers (or 
> video cards) that accept 48bit data - does anyone know?
> 

Maybe giclee fine art color printers in the $20,000+ range, but I doubt 
it.

A way to improve digital B&W printing is to print in duo-tone or even tri-
tone. There are third party adaption kits for Epson printers to do this.

For slide printers, probably only the high-end Kodak LVT printers support 
48 bit. It uses a laser as source. Other film recorders, like the polaroid 
palette, use a CRT as source and thus have a more limited contrast and 
tonal range.

Ingo

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> Everytime I boot up, my monitor dims as the colour profile kicks in.
>

I fear not. Photoshop installs a tool called "Adobe Gamma Loader"
and I assume PaintShop does something similar.
Such tools do a reinitialization of the Gamma LUT to compensate
the difference between the "real" physical gamma of the monitor
and the Windows default gamma of 2.2 (this explains the change
in brightness you may see) . Bus this has little to do with color
management and nothing with ICM.

-Ive

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On Wed, 10 Mar 2004 21:18:28 +0100, "Ive" <ive### [at] lilysoftcom> wrote:

>
>> Everytime I boot up, my monitor dims as the colour profile kicks in.
>>
>
>I fear not. Photoshop installs a tool called "Adobe Gamma Loader"
>and I assume PaintShop does something similar.

I know. It's the Loader doing its job.

PSP has something similar when it starts - it's form of color
management - but I always found it too warm/yellow.

>Such tools do a reinitialization of the Gamma LUT to compensate
>the difference between the "real" physical gamma of the monitor
>and the Windows default gamma of 2.2 (this explains the change
>in brightness you may see) . Bus this has little to do with color
>management and nothing with ICM.

Really? You see this is what always confuses me. I still habe not
totally  fathomed out the differences in color management, profiles,
proofing etc.

>
>-Ive
>

--------------------------------
My First Subgenius Picture Book:
http://www.imbjr.com

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laurent.artaud[AT]free.fr" <"laurent.artaud[AT]free.fr wrote:
> > One thing that I do notice is that if I turn my amp up to full while
> > connected to my 2496 soundcard I hear no hiss or noise at all.
> > Switch it to my SoundBlaster Live and wham! there is so much hiss,
> > even when all outputs are muted in windows ;-)
> >
>
> By the way, which 2496 sound card do you have?

Midiman Delta Dio

I have just done some tests on my system...

First off, the background noise level of my soundcard/amp/speakers is
about -93dB with the Midiman and about -65dB with the SBLive.  I have to
turn my volume control up to at least double what I normal use for music to
be able to hear the -80dB.  I suspect if burnt the WAV to a CD a played it
without my PC on and no other sounds I would be able to hear it at normal
listening volumes.

Next the square wave test.  I made a file split into two halves, all at
24/96.  The first half was just a pure 10kHz note, the second half was the
same but with a 30kHz note added with an amplitude of 1/3 (ie a square wave
filtered off at 48kHz).  Could I tell the difference?  Not a dicky bird,
even though the waveform looked quite different.

I even added harmonics at 20, 30 and 40kHz and could still tell no
difference from the pure 10kHz sine wave.

After a bit more testing, I found I could hear frequencies up to about
18.5kHz, so for me, a sampling rate of 41kHz would sound the same as
anything higher (assuming 10% bandwidth for non-perfect analogue filters).

This has all got rather OT :-)

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