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DVD Audio or SACD ?, Support audio
cyberVera
post Jun 19 2005, 00:52
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QUOTE (listen @ Jun 18 2005, 05:45 PM)
Remember maths class when you find that there is only one parabola that can fit through 3 given points? Well PCM is working on this principal, using the sampled points to guide a continuous waveform. It is -not- the jagged and "digital" (but good enough) approximation that is often described.
*


Right. The only problem are "sampled points" and there 16-bit distance wink.gif. All the rest is fine smile.gif.

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Pio2001
post Jun 19 2005, 02:21
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If they think that louder is better, why would they make CD louder if they want SACD to sound better ??

QUOTE (2Bdecided @ Mar 29 2005, 01:44 PM)
filtering at 19.5kHz was audible to one person who took part in those tests!
*


Filtering at 28 kHz was audible for this person : http://www.hydrogenaudio.org/forums/index....ndpost&p=198806

QUOTE (WmAx @ Mar 28 2005, 01:45 AM)
at this point no confirmed perceptual tests have concluded that anything exceeding a [1]16kHz bandwidth is needed to be transparent for musical program playback for humans.
*


What about this one ?

Yoshikawa, Shokichiro; Noge, Satoru; Yamamoto, Takeo; Saito, Keishi
Does High Sampling Frequency Improve Perceptual Time-Axis Resolution of Digital Audio Signal?
Preprint 4562 in
http://www.aes.org/publications/preprints/search.cfm
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Pio2001
post Jun 19 2005, 02:24
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QUOTE (Pio2001 @ Jun 19 2005, 03:21 AM)
What about this one ?


Oops, sorry, it doesn't deal with musical program unsure.gif
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WmAx
post Jun 19 2005, 04:00
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QUOTE (Pio2001 @ Jun 18 2005, 09:21 PM)
If they think that louder is better, why would they make CD louder if they want SACD to sound better ??


Excellent question. Now, who has an answer? ohmy.gif



QUOTE
Filtering at 28 kHz was audible for this person : http://www.hydrogenaudio.org/forums/index....ndpost&p=198806


I should have been more specific. 1st, I did ammend the 16kHz part of my statement in this thread already, because an unusual sample(with a strong >16kHz constant component at high amplitude, which I suspect is basicly as audible as a sine wave, in the confined circumstances) was easily ABXed, by basicly anyone, that was provided by a previous poster. Fear of this type of sample was probably one of the reasons that the designers of RBCD decided to go ahead and leave room for error, and have a bandwidth >20kHz. As for the person who ABXed a sample with 28kHz filtering, you realize this is a questionable result, especially since this person stated they could not even hear an 18kHz sine wave. I already participated in the thread linked, and we(you and I) already discussed the requirements for a test to have high credibility/reliability. I don't see a need for us to rehash that in this current thread.

QUOTE
What about this one ?

Yoshikawa, Shokichiro; Noge, Satoru; Yamamoto, Takeo; Saito, Keishi
Does High Sampling Frequency Improve Perceptual Time-Axis Resolution of Digital Audio Signal?
Preprint 4562 in
http://www.aes.org/publications/preprints/search.cfm
*


1. I have not reviewed the paper, so I can't comment on the test specifically(do you have a copy you could e-mail me? It would be appreciated if you could. ).

2. As you already stated in the last reply, the test did not use musical program material.

-Chris

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cabbagerat
post Jun 19 2005, 08:22
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QUOTE (Erich w/ an h @ Jun 18 2005, 05:10 PM)
ok, so, the debate rages on between formats... ive a question then.
What would be the idea way to try to represent analogue sound in a digital medium? How can we duplicate, with optimal results, real sound in digital format?
*

I don't understand the distinction you are making between "real sound" and "digital sound". When an analog audio signal is converted into PCM (CD and DVD) it is sampled (limiting it's bandwidth) and quantized (limiting it's signal to noise ratio). If we do the sampling process at 96kHz (with correct filtering) the sampled signal can be used to perfectly reconstruct the original signal up to 48kHz. The quantization step reduces the signal to noise ratio, effectively adding wide-band quantization noise to the signal. For 24bit, the quantization noise is at -144dB (this can be improved in the audible band by noise shaping).

Converting back from PCM to analog gives us our input signal, bandlimited and with a small amount of added wide band noise. The output of the DAC is "real sound" - it's not been tainted by it's trip to the world of PCM in any way other than the bandlimiting and noise (both of which should be inaudible for 24/96). Real-world DACs and ADCs do mangle the sound, but decent ones hardly mangle it at all. What do you base your distinction between "real sound" and "digital sound" on?
QUOTE (cyberVera @ Jun 18 2005, 03:21 PM)
SACD is the first step to do that.
You can find technical info regarding SACD's coding in the web.
*

What makes you believe that DSD is any more "analog" than L-PCM? I suggest you read up on some of the research into the properties of DSD, starting with "Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications" by Lipshitz and Vanderkooy (AES 110th convention). Please go back and read some of the previous discussions of this topic here on HydrogenAudio. The search function will guide you on your way.


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Erich w/ an h
post Jun 19 2005, 08:30
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QUOTE (cabbagerat @ Jun 19 2005, 03:22 AM)
I don't understand the distinction you are making between "real sound" and "digital sound". When an analog audio signal is converted into PCM (CD and DVD) it is sampled (limiting it's bandwidth) and quantized (limiting it's signal to noise ratio). If we do the sampling process at 96kHz (with correct filtering) the sampled signal can be used to perfectly reconstruct the original signal up to 48kHz. The quantization step reduces the signal to noise ratio, effectively adding wide-band quantization noise to the signal. For 24bit, the quantization noise is at -144dB (this can be improved in the audible band by noise shaping).

Converting back from PCM to analog gives us our input signal, bandlimited and with a small amount of added wide band noise. The output of the DAC is "real sound" - it's not been tainted by it's trip to the world of PCM in any way other than the bandlimiting and noise (both of which should be inaudible for 24/96). Real-world DACs and ADCs do mangle the sound, but decent ones hardly mangle it at all. What do you base your distinction between "real sound" and "digital sound" on?


my thought of real sound is sound pre-recording; sound that has yet to be imprinted on a medium. Digital sound is sound that has been put into a digital form; 1s and 0s that when processed acordingly gives us American Pie.

... If digital sound and the sound prior to being digitized is, as you say, a "[perfect] reconstruct [of] the original signal", then why is there a debate at all, and why are different formats still being tested and released? If there was a way to reconstruct the original signal 100%, why are we even discussing anything else?

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Pio2001
post Jun 19 2005, 11:35
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QUOTE (WmAx @ Jun 19 2005, 05:00 AM)
As for the person who ABXed a sample with 28kHz filtering, you realize this is a questionable result


Yes. I was rather answering 2bdecided. From my point of view, this result is not a proof in itself, but it justifies taking some time to test the same sample.

QUOTE
(do you have a copy you could e-mail me? It would be appreciated if you could. ).
*


No. I'd like to read it too, but I don't know if I shall spend 20 $ for it.
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guruboolez
post Jun 19 2005, 13:18
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QUOTE (WmAx @ Mar 28 2005, 12:45 AM)
at this point no confirmed perceptual tests have concluded that anything exceeding a [1]16kHz bandwidth is needed to be transparent for musical program playback for humans.
*


I could try to do it if needed. I know that I'm sensitive to 16KHz lowpassing with some instruments (harpsichord, cymbals...), which appeared as dull, desaturated with ~16Khz lowpass.
I seriously don't think that 16KHz could be fully transparent, even with 'usual music'. At least not with headphones.
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cyberVera
post Jun 19 2005, 13:55
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QUOTE (guruboolez @ Jun 19 2005, 06:18 AM)
QUOTE (WmAx @ Mar 28 2005, 12:45 AM)
at this point no confirmed perceptual tests have concluded that anything exceeding a [1]16kHz bandwidth is needed to be transparent for musical program playback for humans.
*


I could try to do it if needed. I know that I'm sensitive to 16KHz lowpassing with some instruments (harpsichord, cymbals...), which appeared as dull, desaturated with ~16Khz lowpass.
I seriously don't think that 16KHz could be fully transparent, even with 'usual music'. At least not with headphones.
*



The same with me. While being a student, I tested my hearing in a lab and was able to hear up to 19kHz generated noise. Although, do not know what I can now. smile.gif
Besides, ears is not the only part of our body we percept sounds with. If I can not hear frequencies above 20kHz it does not mean that I do not percept them and that they do not influence me.
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cabbagerat
post Jun 19 2005, 15:37
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QUOTE (Erich w/ an h @ Jun 18 2005, 11:30 PM)
... If digital sound and the sound prior to being digitized is, as you say, a "[perfect] reconstruct [of] the original signal", then why is there a debate at all, and why are different formats still being tested and released? If there was a way to reconstruct the original signal 100%, why are we even discussing anything else?
*

There are several reasons new digital formats are being developed. First, with high rate PCM (DVD-A) there are some technical advantages - 24/96 is better than 16/44.1. Whether that difference is audible is still something that is under discussion. Next, it is easier to build DACs and filters which exhibit closer to ideal behaviour in the audio band when given high-rate data. This makes equipment cheaper and better. DSD (as used in SACD for example) is also very easy to build good DACs for. I believe the industry push towards digital path amplifiers (S-Master and friends) has been a major factor in some company's backing of the format.

Lastly, an most important in my mind, is that the new formats offer DRM to music lables. Whether you think this is a good idea or not, it is in the interests of music lables to limit your freedom as much as possible when it comes to your use of their recordings. In my opinion, the next generation of digital formats has been driven by two factors - 5% technical considerations and 95% business considerations.


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Erich w/ an h
post Jun 20 2005, 04:15
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QUOTE (cabbagerat @ Jun 19 2005, 10:37 AM)
Lastly, an most important in my mind, is that the new formats offer DRM to music lables. Whether you think this is a good idea or not, it is in the interests of music lables to limit your freedom as much as possible when it comes to your use of their recordings. In my opinion, the next generation of digital formats has been driven by two factors - 5% technical considerations and 95% business considerations.
*


oh of course. I disagree with the premise of their MO, but I understand it completely... and thats been the case since tinpan ally, only getting worse.

oh well.
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KikeG
post Jun 20 2005, 09:12
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I've added to my web page ( http://www.kikeg.arrakis.es/lowpass/ ) that other sample where a 16 KHz lowpass is audible. It's the Androgyny sample. The difference is not as obvious as in the Chenoa sample, but for me it's still pretty easy to abx (16/16 in 45 sec.)

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WmAx
post Jun 20 2005, 09:49
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QUOTE (Pio2001 @ Jun 19 2005, 06:35 AM)
QUOTE (WmAx @ Jun 19 2005, 05:00 AM)
As for the person who ABXed a sample with 28kHz filtering, you realize this is a questionable result


Yes. I was rather answering 2bdecided. From my point of view, this result is not a proof in itself, but it justifies taking some time to test the same sample.

QUOTE
(do you have a copy you could e-mail me? It would be appreciated if you could. ).
*


No. I'd like to read it too, but I don't know if I shall spend 20 $ for it.
*


I have a friend who has an AES membership(he can download PDFs for free). I'll ask him if he'll let me have a copy. If I can get him to do it, I'll forward you a copy if you give me an e-mail address.

-Chris

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Defsac
post Jun 20 2005, 09:59
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QUOTE (WmAx @ Jun 20 2005, 06:49 PM)
I have a friend who has an AES membership(he can download PDFs for free). I'll ask him if he'll let me have a copy. If I can get him to do it, I'll forward you a copy if you give me an e-mail address.

-Chris
*

It says on the site it'll still cost him $5.

QUOTE
Single Convention Preprints and Conference Papers are $5.00 for AES Members and $20.00 for non-Members.


I have a copy of the document but it says that you need "direct permission from the Journal of the Audio Engineering Society" to distribute it.

Edit: I can provide a description of the test without infringing copyright.

The sample material used for the test was recorded on a 96 kHz 16 bit DAT tape. The reference sample (sample R) had a FIR lowpass filter at 40 kHz while the test sample (sample X) had a lowpass at 20 kHz. 11 adult males, 22-24 years of age, were asked to identify which sample was X and which was R. The result was statistically significant (pval < 0.05), and the folllowing was concluded.

QUOTE
From the experiment using a pulse train signal, it was suggested that widening the frequency range of the audio system improves perceptual time-axis resolution. Sound recording and reproduction system using a high sampling (i.e. 96 kHz sampling) will be helpful from the view point of improving time-axis resolution of the digital audio signals.


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Nika
post Jun 20 2005, 14:12
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QUOTE (Kees de Visser @ Jun 18 2005, 03:54 PM)
Digital clipping can be shown on a good digital peak-level meter with calibrated overload indication. The number of consecutive clipped samples (full scale) is an indication for the amount of audible clipping. No clipped samples means the audio can be reconstructed with a good DAC.
One or more FS samples indicate clipping, but it doesn't have to be audible.
I'd gladly check your files for digital clipping and even try to remove the clipping if you like. The music will be softer then, but that's exactly the mastering engineer's dilemma.
*


I haven't read far enough ahead in the thread to see if this was addressed. The above statement is not correct. For more read the paper on the consequences of traditional peak meters here:

http://www.cadenzarecording.com/papers

It is very possible (in fact likely) that signals wherein the samples approach FS will clip the output of even a good D/A converter. The digital reconstruction filters on those D/As simply can't reconstruct above FS even if the analog can.

Cheers!
Nika
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Nika
post Jun 20 2005, 14:23
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QUOTE (Erich w/ an h @ Jun 19 2005, 01:30 AM)
... If digital sound and the sound prior to being digitized is, as you say, a "[perfect] reconstruct [of] the original signal", then why is there a debate at all, and why are different formats still being tested and released? If there was a way to reconstruct the original signal 100%, why are we even discussing anything else?
*


Erich,

Your question implies that the changes to formats have been made because they inherently provide an improvement. This is an optimistic view of the marketing and multinational corporations today. Sony has proved inside their building that 44.1kS/s is better than DSD and can sound as good for any recording/mixing/delivery situation as higher L-PCM sample rates such as 96kS/s and 192kS/s. There is a reason that you don't hear about this.

96kS/s was originally proposed back in the early 90s or late 80s as a solution to a problem the industry was having with filter design. That problem was fixed long ago, but the push for higher fs had started and it was viewed as potentially profitable as well, so despite the lack of need for it, the industry pushed on.

SACD (DSD) was similarly created as a patch for a problem that the industry had. But while the industry fixed that problem (that being integral and differential non-linearity in converters yielding high distortion) the DSD marketing machine had already gotten started. And therefore, despite the fact that modern converters designed for L-PCM use inherently have a leg up on DSD, Sony can't back down now and DSD continues as a major marketing push.

In each of these situations the industry had a choice: fix what was wrong or develop a new format altogether. Since there is a lot of profitable money tied with new formats we can't be surprised which way the labels and consumer gear manufacturers went, while the R&D guys at the converter manufacturers were busy fixing the problems...

Nika
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KikeG
post Jun 20 2005, 14:23
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QUOTE (Defsac @ Jun 20 2005, 09:59 AM)
Edit: I can provide a description of the test without infringing copyright.

...
*


But, did the test designers take any measures to avoid speaker distortion products falling into audible band, when feed with signals very rich in ultrasonic content, such as the pulse train they make mention of?

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Nika
post Jun 20 2005, 14:29
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QUOTE (cyberVera @ Jun 19 2005, 06:55 AM)
The same with me. While being a student, I tested my hearing in a lab and was able to hear up to 19kHz generated noise. Although, do not know what I can now.  smile.gif
Besides, ears is not the only part of our body we percept sounds with. If I can not hear frequencies above 20kHz it does not mean that I do not percept them and that they do not influence me.
*



1. Was this test using linear phase filtering? That will throw off everything immediately, as non-linear phase filtering will create audible phase problems in the audible range, despite the presence of HF content.

2. Were the speakers linear above 20kHz? Non-linear speakers above 20kHz can create audible bandwidth distortion when presented with HF content. Again, however, this isn't an indication of audibility above 20kHz but rather audibility below.

3. Despite the persistent notion that we can hear with organs other than our ears, absolutely nothing has been presented on the market that substantiates that we can do this for HF content at musical program listening levels. We can "feel" HF content at several hundred dB above normal listening levels (you would certainly "feel" it if someone started sonar-welding your skin, for example) but even studies that have been done on this subject have indicated no "perceptibility" of HF content. Nobody asked the "listeners" in such tests to only indicate the HF content that they could "hear" and to ignore the content that they "felt in other ways." U

Nika
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Nika
post Jun 20 2005, 14:34
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QUOTE (Defsac @ Jun 20 2005, 02:59 AM)
QUOTE
From the experiment using a pulse train signal, it was suggested that widening the frequency range of the audio system improves perceptual time-axis resolution. Sound recording and reproduction system using a high sampling (i.e. 96 kHz sampling) will be helpful from the view point of improving time-axis resolution of the digital audio signals.

*



This type of stuff is really annoying. How in the world did they draw that conclusion from that test?

Nika
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WmAx
post Jun 20 2005, 20:31
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QUOTE (Defsac @ Jun 20 2005, 04:59 AM)
Single Convention Preprints and Conference Papers are $5.00 for AES Members and $20.00 for non-Members.


That is for the old standard membership model. There is an online subscription model that allows for unlimited downloads of papers, which was recently introduced within this year.

QUOTE
I have a copy of the document but it says that you need "direct permission from the Journal of the Audio Engineering Society" to distribute it.


Then, consider me an infringer(of the trade of AES research papers).

As for the general description; that's not exactly what I need. I need to review a paper in full before I come to any conclusion(s). This is not a published paper, for one thing, so the quality of paper could be anything.

-Chris

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WmAx
post Jun 20 2005, 20:43
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There is a good explanation of the behaviour of DACs when in the presence of levels nearing 0dBfs and the relation to peak output distortion(s), in this paper:

OdBFS+ Levels in Digital Mastering
Sgren H. Nielsen and Thomas Lund
AES Preprint 5251

This paper(unlike most) is available for free download from the authors:
http://www.tcelectronic.com/media/nielsen_...00_0dbfs_le.pdf

-Chris
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Pio2001
post Jun 20 2005, 22:45
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QUOTE (KikeG @ Jun 20 2005, 03:23 PM)
But, did the test designers take any measures to avoid speaker distortion products falling into audible band, when feed with signals very rich in ultrasonic content, such as the pulse train they make mention of?
*


Good question. While this might have little effect for musical content, the difference can be very big with such signals. I remember, back in school, having played with a generator with a very cheap speaker plugged in (the kind of speaker for mono TV set or FM-Alarm-clock, if you see what I mean), and the 10 kHz sine, square and triangle had a completely different sound because of distortion.
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Defsac
post Jun 20 2005, 23:06
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QUOTE (KikeG @ Jun 20 2005, 11:23 PM)
But, did the test designers take any measures to avoid speaker distortion products falling into audible band, when feed with signals very rich in ultrasonic content, such as the pulse train they make mention of?
*

If they did it wasn't mentioned in the preprint.
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batagy
post Jun 21 2005, 09:46
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QUOTE (coastalbumm @ Jun 8 2005, 11:58 PM)
Listen to the remastered Dark Side of The Moon on SACD/CD Hybrid, it was engineered by the ORIGINAL engineer
*

Hi,
Just for correction, the engineer for the new Dark Side Of The Moon SACD was James Guthrie, and the ORIGINAL engineer was Alan Parsons, so that was NOT engineered by the original engineer. wink.gif
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Kees de Visser
post Jun 23 2005, 19:53
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QUOTE (Nika @ Jun 20 2005, 03:12 PM)
QUOTE (Kees de Visser @ Jun 18 2005, 03:54 PM)
...No clipped samples means the audio can be reconstructed with a good DAC...
*


I haven't read far enough ahead in the thread to see if this was addressed. The above statement is not correct. For more read the paper on the consequences of traditional peak meters here:

http://www.cadenzarecording.com/papers

It is very possible (in fact likely) that signals wherein the samples approach FS will clip the output of even a good D/A converter. The digital reconstruction filters on those D/As simply can't reconstruct above FS even if the analog can.

Cheers!
Nika
*



Hi Nika,
since reconstruction filter "headroom" isn't part of any DAC specification I know of, it's hard to know the performance in that respect. After I read your post I called Prism (my DAC brand) to ask about their specs and although they couldn't give numbers ("would have to look that up") they assured me that their reconstruction filter has been designed with "quite some" margin for overloads.
Do you know of any test or (AES?) specification that can help to quantify DAC overload performance ? I'd love to know the limits of my DAC.
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