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32-bit capable DACs, Split from: "WASAPI plug-in version 3.0 beta" (94907)
IgorC
post May 26 2012, 22:46
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QUOTE (naturfreak @ May 26 2012, 18:26) *
It ist even worse in the case microphones are involved, because the output signal of a average microphone without amplifier is only around -40 dBm, but the noise level is at around -130 dBm. Therefore 90 dB S/N is the best you can get in this case.
Amplifers produce additional noise. You simply can't lower the level of noise by using amplifiers.

Ouch You have just messed up dBm and dB again. laugh.gif

The noise level in a good silent studio is around (well, it's already pretty ideally) -120 dB, not -120 dBm. Because -120dBm is much less than -120dB (-120 dBm = -150dB -> it's too low for real-life noise).

This post has been edited by IgorC: May 26 2012, 22:57
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naturfreak
post May 26 2012, 23:11
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I'm not talking about the level auf acoustic noise in the studio or atmosphere, Igor. I'm talking about the noise level in electrical parts, wires, circuits.
The difference between -130 dBm and -40 dBm is 90 dB, because the in the calculation of a difference of levels the reference level of 1 mW will be discarded.
So the limitation of S/N is in the electrical domain, not in the possible difference in air pressure.
You can shift the level of loudness, yes, but in the electrical domain S/N will stay the same.
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IgorC
post May 26 2012, 23:31
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QUOTE (naturfreak @ May 26 2012, 19:11) *
The difference between -130 dBm and -40 dBm is 90 dB, because the in the calculation of a difference of levels the reference level of 1 mW will be discarded.
So the limitation of S/N is in the electrical domain, not in the possible difference in air pressure.
You can shift the level of loudness, yes, but in the electrical domain S/N will stay the same.

Yes,S/N will remain the same. It's true.

But there is no ADC with such low noise -130 dBm (-160dB). -130 dB maybe. -130 dBm - science fiction.

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SoNic67
post May 26 2012, 23:44
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QUOTE (IgorC @ May 26 2012, 15:03) *
QUOTE (SoNic67 @ May 26 2012, 15:23) *
For real audio resolution, the number to look for is the THD+N.

Can You defend this statement with some additional information?

Then why include Dynamic Range into specification? Just go for THD+N, right?

The final product is the music. Whatever is not music should be cosidered as altering the original signal. Therefore, for real resolution, you need to look at THD+N numbers. S/N is just the "N" part of that sum. Unless you really like the specific distortions... case for many tube-lovers.
Anyway I am saying that the physically limit of the human hearing is around 21-22 dB. Even the thermal noise of the air molecules in a 25C room is at that level. There is absolutelly no point for more than 24 bit (at 96kHz) reproduction of audio signals destined to be heard by humans.

DR, as is defined, is kind of a marketing tool. Is the SNR measured at -60dB of the full scale and at the value obtained, add 60dB. Well, that tells how good is the system at low level reproduction, but the distortions are really creeping up at higher levels, close to 0dB. Is was a desperate attmpt to hide the glitching distortion in early multibit DAC's. Not usefull today when all the DAC's have a zero-detection circuit that mutes the output when "zero" PCM code is detected.

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naturfreak
post May 27 2012, 00:18
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QUOTE (IgorC @ May 26 2012, 23:31) *
But there is no ADC with such low noise -130 dBm (-160dB). -130 dB maybe. -130 dBm - science fiction.

Yes, there is no ADC with this low noise level and the consequence is there is no point designing or producing ADC oder DAC that uses more than 24 Bit of (integer) sample depth.
Are there ADC or DAC using 32 bit floating-point for conversation or internal format?

QUOTE (SoNic67 @ May 26 2012, 23:44) *
Anyway I am saying that the physically limit of the human hearing is around 21-22 dB. Even the thermal noise of the air molecules in a 25C room is at that level. There is absolutelly no point for more than 24 bit (at 96kHz) reproduction of audio signals destined to be heard by humans.

I agree with you.
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SoNic67
post May 27 2012, 03:11
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As for ADC reality:
In my E-MU 1820m I have one of the best ADC around, same like in ProTools professional mastering gear: AK5394A. That is capable of "only" THD+N=-110dB... True 18 bits.
Higher level than that there is the DAD-AX24 with THD+N at -115dB (true 19bits) for 10 times the price.

From my findings, many of the old recordings are at a very poor quality comparativelly, derived from the original master analog tapes. Even if they are sold in DVD-A, SACD or digital 24/96 formats, they are actually below the capabilities of 16 bit and with max bandwidth limited often at 17- 18kHz...
Newer recordings have bandwidth brickwalled at 20kHz and are compressed in such a manner that the true dynamic is only at 12-13 bit resolution level (at best).

This post has been edited by SoNic67: May 27 2012, 03:15
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2Bdecided
post May 28 2012, 14:21
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QUOTE (Jackal29a @ May 26 2012, 19:31) *
I'm quite surprised by some people's reactions, it has been stated here that both ASIO and WASAPI provide no advantage whatsoever over DS therefore being as pointless as 32bit output so, why not ditch these plugins altogether? the same could possibly said about FLAC, APE or any other lossless codec that provides no audible advantage over well encoded 192Kbps, and so on and so forth.
Because that difference is sometimes audible, and always measurable. Whereas 24-bits vs more than 24-bits is, AFAIK, completely and utterly undetectable anywhere in the analogue audio world.

Cheers,
David.
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2Bdecided
post May 28 2012, 14:25
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QUOTE (IgorC @ May 26 2012, 22:46) *
The noise level in a good silent studio is around (well, it's already pretty ideally) -120 dB
That's useful! "-120dB" means "one million times smaller than" - so "the noise level in a good silent studio is around one millions times smaller than..." what?! wink.gif

FWIW I don't know of any studios that are sound proofed down to 0dB SPL across the audible range (around 20dB SPL is a common target, rarely reached across the board) - but even if they exist, you'd better hope that the musicians don't ruin it by breathing.

Cheers,
David.
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Roseval
post May 28 2012, 17:45
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I wonder if the “32 bit DAC” has anything to do with dynamic range.
ESS Sabre is advertised as a 32 bit “DAC” simply because they do some DSP (probably ASRC to reduce jitter) and they do so in 32 bit float hence the 32 bit.

Audio editors often allows you to store the project in float.
You don’t want a quantization error each time you save your work.
Can imagine it is convenient to feed this straight into a DAC

In the computer world 24 is a weird number.
8/16/32/64/128 are the common word length.
Might be convenient to keep your audio in 32 or 64 when processing.


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IgorC
post May 28 2012, 18:14
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QUOTE (2Bdecided @ May 28 2012, 10:25) *
FWIW I don't know of any studios that are sound proofed down to 0dB SPL across the audible range (around 20dB SPL is a common target, rarely reached across the board) - but even if they exist, you'd better hope that the musicians don't ruin it by breathing.

Your reaction is to expect. But if You look into the context of this particular discussion and I've mentioned that those are ideal (pretty unrealistic) numbers.
And yes, those SPL numbers are available even on wiki for easy reading. I was perfectly aware of them. Thank You.

The message was the other.

This post has been edited by IgorC: May 28 2012, 18:15
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saratoga
post May 28 2012, 19:02
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QUOTE (Roseval @ May 28 2012, 12:45) *
Audio editors often allows you to store the project in float.
You don’t want a quantization error each time you save your work.
Can imagine it is convenient to feed this straight into a DAC


32 bit float has only a 23 bit mantissa, so I tend to think most stuff using floating point uses doubles.

Anyway, DACs take integer precision, not fp, so whatever you are working with is irrelevant. Its going to be cast to whatever packed integer format the DAC specifies. Its not like your software even knows that though, the driver keeps the DAC and your programs quite separate.
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db1989
post May 28 2012, 19:10
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QUOTE (Jackal29a @ May 26 2012, 19:31) *
I'm quite surprised by some people's reactions, it has been stated here that both ASIO and WASAPI provide no advantage whatsoever over DS therefore being as pointless as 32bit output so, why not ditch these plugins altogether?
No audible advantage. (Disregard broken software or hardware.) But they exist exactly to provide their other advantages, such as low latency, exclusive access, etc. This has been gone over many times.

QUOTE
the same could possibly said about FLAC, APE or any other lossless codec that provides no audible advantage over well encoded 192Kbps, and so on and so forth.
Even if the above were not misunderstood, this would not follow from it. Lossless archiving has objective advantages completely separate from any discussion of perceptual transparency: archival and future-proofing are the main ones I can offer now, but that’s not all.
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Canar
post May 28 2012, 19:19
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QUOTE (db1989 @ May 28 2012, 11:10) *
(Disregard broken software or hardware.)
No need to totally disregard, either, as this is one of the other valid reasons for using these outputs. I found WASAPI output in a Windows VirtualBox to be more reliable than DS.


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Kohlrabi
post May 28 2012, 21:52
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QUOTE (Canar @ May 28 2012, 20:19) *
I found WASAPI output in a Windows VirtualBox to be more reliable than DS.

And I experienced the same in my wine+foobar2000 endeavours.


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Rotareneg
post May 29 2012, 03:23
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But I needed a DAC that could accurately play back a recording of a mouse farting during a space shuttle launch! sad.gif
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2Bdecided
post May 29 2012, 09:35
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QUOTE (IgorC @ May 28 2012, 18:14) *
The message was the other.
Oh, I thought you were telling someone they weren't using "dB" correctly, before giving an example that was incorrect by a factor of one million. wink.gif
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Canar
post May 29 2012, 16:28
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QUOTE (Rotareneg @ May 28 2012, 19:23) *
But I needed a DAC that could accurately play back a recording of a mouse farting during a space shuttle launch! sad.gif
But from precisely how many yards are you recording the mouse fart? It changes everything!


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greynol
post May 29 2012, 18:37
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For me it didn't matter. I had the volume turned off on my TV when I watched the launch.

Forget that the mouse was already in my cat's stomach.


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IgorC
post May 30 2012, 01:34
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QUOTE (2Bdecided @ May 28 2012, 10:25) *
but even if they exist, you'd better hope that the musicians don't ruin it by breathing.

Breathing can be part of recorded program. So your treament of it like some unwanted noise is wrong.

QUOTE (2Bdecided @ May 29 2012, 05:35) *
Oh, I thought you were telling someone they weren't using "dB" correctly, before giving an example that was incorrect by a factor of one million. wink.gif

Overplayed.
Show me where my statement was by a factor of one million time wrong.

I think You can apply now your criticism to your very own post. smile.gif

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Arnold B. Kruege...
post May 30 2012, 14:17
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QUOTE (Rotareneg @ May 28 2012, 22:23) *
But I needed a DAC that could accurately play back a recording of a mouse farting during a space shuttle launch! sad.gif


BTW, got any idea what the output of the finest microphone in the world would be like under those circumstances?

Not pretty!
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Arnold B. Kruege...
post May 30 2012, 14:23
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QUOTE (IgorC @ May 26 2012, 18:31) *
QUOTE (naturfreak @ May 26 2012, 19:11) *
The difference between -130 dBm and -40 dBm is 90 dB, because the in the calculation of a difference of levels the reference level of 1 mW will be discarded.
So the limitation of S/N is in the electrical domain, not in the possible difference in air pressure.
You can shift the level of loudness, yes, but in the electrical domain S/N will stay the same.

Yes,S/N will remain the same. It's true.

But there is no ADC with such low noise -130 dBm (-160dB). -130 dB maybe. -130 dBm - science fiction.


Take a DAC with -130 dB noise referenced some higher level than 0 dBm, add a properly designed resistive divider with maybe an op amp buffer, and viola, -130 dBm noise levels.

I believe that there are now commercial DACs with -130 dB FS noise - ESS Sabre DACs.

There are several other DACs that come close Probably very little but money keeps you from connecting a bunch of them in parallel in such away that the noise drops -3 dB every time you double the number of DACs connected in parallel.
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Arnold B. Kruege...
post May 30 2012, 14:29
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QUOTE (SoNic67 @ May 26 2012, 18:44) *
DR, as is defined, is kind of a marketing tool.


No, it is a good spec, but yes it is incomplete if you want to know what is going on in all relevant conditions. That's why spec sheets have more than one number on them!

QUOTE
Is the SNR measured at -60dB of the full scale and at the value obtained, add 60dB. Well, that tells how good is the system at low level reproduction,


Which is a good thing to know, no?

QUOTE
but the distortions are really creeping up at higher levels, close to 0dB.


Agreed, so we have other specs to characterize them. We expect to see more than one number!

QUOTE
Is was a desperate attmpt to hide the glitching distortion in early multibit DAC's.


No, it was an effective attempt to expose low level glitching which was far more common before Delta-Sigma converters became the rule.

QUOTE
Not usefull today when all the DAC's have a zero-detection circuit that mutes the output when "zero" PCM code is detected.


There is no such thing in general. You need to study up on how modern converters work.
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2Bdecided
post May 31 2012, 10:24
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QUOTE (IgorC @ May 30 2012, 01:34) *
Show me where my statement was by a factor of one million time wrong.
Quoting a real world sound level as "-120dB".

No units / 1M times wrong in the implied units - take your pick - a physics teacher would fail you either way.


Which isn't the kind of thing I'd usually be so uncharitable as to criticise - but you were laughing at naturfreak for a much smaller error.


I do agree we don't need too many bits. I can hear breathing perfectly well on 16-bit recordings.

Cheers,
David.
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IgorC
post May 31 2012, 10:59
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QUOTE (2Bdecided @ May 31 2012, 06:24) *
Quoting a real world sound level as "-120dB".

I've mention it with that reference level because if not it would just confuse more Naturefreak. See discussion first.



QUOTE (2Bdecided @ May 31 2012, 06:24) *
No units / 1M times wrong in the implied units - take your pick - a physics teacher would fail you either way.

A physics teacher would fail you teaching that different measurements can mean the same with different reference levels.
And I've already mention why I did it this way.

You pick a wrong person here to teach about sound pressure levels and unit of measurements. wink.gif Wrong person, wrong topic.

QUOTE (2Bdecided @ May 31 2012, 06:24) *
but you were laughing at naturfreak for a much smaller error.

The idea was to make soft a correction. But I can see that smile icon was interpreted some aggressive way of laughing out.

This post has been edited by IgorC: May 31 2012, 11:12
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SoNic67
post Jun 1 2012, 01:50
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QUOTE (Arnold B. Krueger @ May 30 2012, 09:29) *
There is no such thing in general. You need to study up on how modern converters work.

Now you upset me...
Want examples? AK4683, page 1 "Zero Detect Function". PCM1794, page 1 "Zero Flag" (mute external or internal "soft mute"). CS4398, pags 21 explains the "Mute Control": "use of the Mute Control function can enable the system designer to achieve idle channel noise/signal-to-noise ratios which are only limited by the external mute circuit". All the sigma-delta DAC's will output HF garbage if not muted, as opposed to old-school AD1955 page 1 specifies that SNR is 120dB "not muted".

THD+N is the ONLY realistic number that tells the accuracy of audio output. SNR is just a part of that number.

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