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44.1 vs 88.2 ABX report at AES
Arnold B. Kruege...
post Jul 17 2010, 20:35
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QUOTE (greynol @ Jul 17 2010, 13:33) *
QUOTE (Arnold B. Krueger @ Jul 16 2010, 22:39) *
QUOTE
The reconstruction filter is necessarily analog.

Yes, but it is designed for the oversampling frequency. This puts it octaves above the Nyquist frequency for the wave being converted.

Are you trying to say that the analog low pass filter at the end of the DAC on my old 8x oversampling CD player has a corner frequency at somewhere around 160 Hz?


Not at all, unless there was a typo above and you meant 160 KHz, not the 160 Hz we see abpve. 160 KHz is a probable number, if a little high.

What was the Nyquist frequency for the sample rate of the digital data that your old CD player reconstructed - 44.1 KHz, right?

What would *octaves* above 44.1 KHz be? Well at least 160 KHz (2 octaves).

You said 8x, right? So the oversampling frequency is about 320 KHz, and the corner frequency of the low slope analog reconstruction filter for 320 KHz might be in the 60 KHz - 160 KHz range.

These days the analog reconstruction filter is implemented right on the DAC chip, but in times past (maybe 5 years ago) it was outboard. Its corner frequency could be calculated from schematics in the application notes. Sometimes the AN stated it explicitly. For example, the AD1853 AN shows an analog reconstruction filer with a -3 dB point of 75 KHz and a third order Gaussian characteristic. The AD1853 AN shows some very interesting broadband plots of the response of the digital filter that provides the brick wall filtering.

AD 1853 AN
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greynol
post Jul 17 2010, 20:48
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QUOTE (Arnold B. Krueger @ Jul 17 2010, 12:35) *
160 KHz is a probable number, if a little high.

How can it be a little high when it is already lower than what you specify as to how these things are done?

So please, tell us, what is the point in a CD player having a lowpass at 160 kHz when the audio will never be above 22.05 kHz?

Sorry Arnold, but I don't think you know what you're talking about.

This post has been edited by greynol: Jul 17 2010, 21:12


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AndyH-ha
post Jul 17 2010, 22:02
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I also may not know what I'm talking about, but that high frequency cutoff seems quite in line with what the whole purpose of oversampling is about -- moving the alias image way, way up above the audio band so it is easy to filter off. Since 30kHz, for example, is already well above the audio band, 160kHz could be considered a bit higher than necessary to adequately preserve the audio band.

QUOTE
AndyH-ha, that might be on purpose. For two lowpass filters of identical computational complexity, with A being allowed to inject more imaging above f than B, et ceteris paribus, A has potentially higher passband performance (0-f Hz) than B

I have no idea if it is a design tradeoff or an intrinsic, unavoidable aspect of sigma-delta. I've never seen it mentioned in print. My point is that it is a real difference in results between sampling at 44.1kHz vs sampling at 88.2kHz or 96kHz (although the actual sample taking is at 2.82MHz vs 5.64MHz).

It may be true, as Arnold stated, that there is no difference in final results between downsampling to 44.1kHz from 88.2kHz vs from 96kHz, but the process is surely different. It takes much longer for CoolEdit to dowwnsample from 96kHz.

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greynol
post Jul 17 2010, 22:13
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The reason for oversampling in old CD players is as you said, to push the images far enough out so that one can good results using a much simpler and cheaper reconstruction filter. Where is the corner frequency of this filter? Far closer to 22 kHz than 160 kHz. If Arnold takes issue with this then I'll direct him to speak with all the professors I had at the university where I learned about digital communications and signal processing. Although I've forgotten much since I earned my BS, I do remember this quite vividly.

I noticed that you mentioned aliasing. It's and issue when it comes to initial sampling and down-sampling, not with reconstruction. Aliasing is when images overlap; it is not the name of the image.

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Arnold B. Kruege...
post Jul 17 2010, 23:04
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QUOTE (greynol @ Jul 17 2010, 15:48) *
QUOTE (Arnold B. Krueger @ Jul 17 2010, 12:35) *
160 KHz is a probable number, if a little high.

How can it be a little high when it is already lower than what you specify as to how these things are done?


?????????????

If I said something like that you'll have to quote it to me. I surely had no such intentions.

QUOTE
So please, tell us, what is the point in a CD player having a low pass at 160 kHz when the audio will never be above 22.05 kHz?


The corner frequency of the analog low pass is set high so that it doesn't cause losses in the pass band. It is usually a relatively simple filter with gentle roll-off.

The analog low pass relates to the oversampling frequency, not the sampling frequency of the data from the media.

If you take the time to read the reference I've already cited, they show how this works with detailed spectral analysis.

You must have missed the link the first time I posted it - here it is again: second Link to detail;ed technical data for an oversampling DAC including spectral analysis

This post has been edited by Arnold B. Krueger: Jul 17 2010, 23:04
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Arnold B. Kruege...
post Jul 17 2010, 23:14
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QUOTE (greynol @ Jul 17 2010, 17:13) *
The reason for oversampling in old CD players is as you said, to push the images far enough out so that one can good results using a much simpler and cheaper reconstruction filter. Where is the corner frequency of this filter? Far closer to 22 kHz than 160 kHz. If Arnold takes issue with this then I'll direct him to speak with all the professors I had at the university where I learned about digital communications and signal processing. to initial sampling and down-sampling, not with reconstruction.


I don't get all the smoke and fire about profs and BS.

Before you posted this stuff I had previously provided a referene that explicity says that the analog reconstruction filter of a certain high quality, fairly recent DAC is 65 KHz. This will work for a 44 KHz DAC, and can work for a 96 KHz DAC that is not intended for instrumentation. It is probably inappropriate for a 192 KHz DAC.

Here's the reference for the third time:

AD 1853 DAC data sheet

The specifc note about the corner frequency of the analog filter is on page 15. Figures 10, 14 and 15 show the wideband output of the device, and the spurious responses that the analog filter targets. They aren't much.


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greynol
post Jul 17 2010, 23:27
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I'm taking issue with what I thought was a blanket statement about oversampling and filtering, though in your response to benski, I did see you restated what you said which included qualifiers, which I overlooked.

As it concerns my CD player and your notion that the corner frequency of the lowpass is around 160 kHz, the smoke and fire isn't coming from my profs. wink.gif

This post has been edited by greynol: Jul 18 2010, 00:50


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Arnold B. Kruege...
post Jul 18 2010, 01:52
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QUOTE (greynol @ Jul 17 2010, 18:27) *
I'm taking issue with what I thought was a blanket statement about oversampling and filtering, though in your response to benski, I did see you restated what you said which included qualifiers, which I overlooked.

As it concerns my CD player and your notion that the corner frequency of the lowpass is around 160 kHz, the smoke and fire isn't coming from my profs. wink.gif


Here's another data sheet, this time for the AD 1955:

AD 1955 data sheet

Figure 7 page 18 shows an analog filter -3 dB point of 100 KHz.

BTW, your original number for the corner frequency of this filter was 160 Hz...
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amandinepras
post Jul 19 2010, 00:52
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Thanks all for your interest in our paper,
I received an invitation from hydrogenaudio to provide further details on our work.
Thus I will do my best to answer a few questions I could extract from the discussion.

- We used Pyramix 6.0 for down-sampling, as this software is currently used by a lot of audio professionals who produce HD recordings.
- Regarding the statistics, the "p" we provided for the results refers to the probability that we got the result by chance. Traditionally for this kind of test (here an ABX), researchers consider that if p<.05, the result is not obtained by chance (as the probability is below 5%), thus participants could discriminate. If .05<p<.1, it may be that the result was not obtained by chance but it's not for sure, that's what is called "a tendency".
If the test was easy, we would not need statistics, as participants would have almost 100% of good answers. But this test was extremely challenging for the expert listeners, implying a lot of errors even if some of them could perceive some differences between formats in specific cases (musical excerpt, type of format comparison).
- There is no proof that upsampling doesn't introduce artifacts.
- Regarding our choice of format comparison and technical chain, our purpose was to investigate perceptive differences between 88.2 vs. 44.1 in "real-life" use of the equipment, thus by taking into consideration what happens in music production and release: in a few cases, music is produced and released in high-resolution (thus playback in high-resolution); in more cases, music is produced in high-resolution and then down-sample into 44.1 for commercial release (thus playback in 44.1); in a lot of cases, music is produced and released in 44.1 (thus playback in 44.1).
We used the Fireface DAC as it was the only one that allowed us to switch sample rate with a reasonable delay for the test (less than 1sec.). I wish we could use a better one. However, the Fireface is still pretty good compared to most playback systems people use in their house.

I am a sound engineer myself and started working in research as a part time job 3 years ago. I was glad to work on the high-resolution project as I have heard a lot of discussions in studios and during my sound recording studies on the topic. My main question was if it was worth working in High-Res when the project was to be released in 44.1.
This AES paper is the first publication for this study and provides a few answers, maybe not enough for most of us. There will be more stuff coming up. And maybe other labs will work on that topic too as they are A LOT of tests to be done.

Bottom line, although the topic is interesting, mainly these days when the Blue Ray Pure Audio is to be defined, never forget that differences between formats, ADC, DAC,... remain extremely subtle compared to differences between miking techniques, room acoustics, and of courses musicians and their instruments!
Best,
Amandine
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greynol
post Jul 19 2010, 00:56
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QUOTE (Arnold B. Krueger @ Jul 17 2010, 17:52) *
Here's another data sheet, this time for the AD 1955:

I see your point, though I don't think it's the one used in my CD player.

QUOTE (Arnold B. Krueger @ Jul 17 2010, 17:52) *
BTW, your original number for the corner frequency of this filter was 160 Hz...

...and you typed "abpve", though I didn't decide to make much of it.


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Arnold B. Kruege...
post Jul 19 2010, 03:35
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QUOTE (amandinepras @ Jul 18 2010, 19:52) *
- There is no proof that upsampling doesn't introduce artifacts.


By stating the question as a negative hypothesis, you've automatically made the proof that you have demanded exceedingly difficult or impossible.

Rather, I'll try to restate the problem in a fair and balanced way: Is it possible to upsample without introducing audible artifacts?

(1) If you believe in the efectiveness traditional audio measurements and known audible thresholds for the things they measure, it is easy to show that there are upsamplers whose added noise and distortion are well below established audible thresholds. I'm talking a safety factor of several orders of magnitude.

(2) If you believe that there are other competent people who know how to do sensitive listening tests, then we have the results of numerous listening tests that show that upsampling can be totally undetectable.
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Arnold B. Kruege...
post Jul 19 2010, 11:35
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QUOTE (amandinepras @ Jul 18 2010, 19:52) *
- We used Pyramix 6.0 for down-sampling, as this software is currently used by a lot of audio professionals who produce HD recordings.


You might want to compare this product's capabilties along these lines by checking out the results posted at

Infinitewave's SRC technical tests

Many of us use Adobe Audition, and its clean processing is one reason why. Audition and a number of other products appear to at least match Pyramix 6 in all areas tested, and beat it in several areas by up to 30 dB. On balance I don't see any flaws in the Pyramix that would necessarily invalidate results obtained by using it. The Infinitewave tests put a number of highly regarded products in an even poorer light.

IME the tools that audio professionals use can have a lot to do with hype and tradition. Perhaps the most widely used and highly regarded DAW software of all time has been Pro Tools, but at times in the past, it had pretty substandard performance.
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Arnold B. Kruege...
post Jul 19 2010, 11:48
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QUOTE (amandinepras @ Jul 18 2010, 19:52) *
We used the Fireface DAC as it was the only one that allowed us to switch sample rate with a reasonable delay for the test (less than 1sec.). I wish we could use a better one. However, the Fireface is still pretty good compared to most playback systems people use in their house.


It's not clear which Fireface DAC that you used as there are several models. I searched around on the web for some technical tests. I found this test of the Fireface 800:

Link to Fireface 800 technical test

These results don't look particularly impressive to me. They appear to be to be representitive of hi rez digital players at the lower end of the market.

I've done many similar tests with this audio interface:

Lynx Two 24/192 technical tests


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googlebot
post Jul 19 2010, 12:19
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QUOTE (Arnold B. Krueger @ Jul 19 2010, 12:48) *
Link to Fireface 800 technical test

These results don't look particularly impressive to me. They appear to be to be representitive of hi rez digital players at the lower end of the market.


Lets not criticize where no critic is due.

In any other context, you would have (rightfully) smashed TOS8, ABX, and your contradicting experience into anyones face, who had claimed that a DAC with the capabilities of the Fireface would not be sufficient for testing and comparing real world music.

The reasons, which have been brought forward for the choice of the Fireface, especially fast sample rate switching and real world setup similarity, are sensible. They wouldn't be, if "real world" had been an excuse for a sub-standard device, but it isn't by any means.

QUOTE (Arnold B. Krueger @ Jul 19 2010, 04:35) *
By stating the question as a negative hypothesis, you've automatically made the proof that you have demanded exceedingly difficult or impossible.

Rather, I'll try to restate the problem in a fair and balanced way: Is it possible to upsample without introducing audible artifacts?


Completely agree.

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Pio2001
post Jul 19 2010, 12:31
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Thanks for joining the discussion, Amandine.

Your work is very interesting. But I think that it is necessary to run all the raw data into a global analysis instead of picking seemingly significant results among all possible comparisons.

I see that you calculated the p values for many different cases. Basically, for the 16 listeners as individuals, then for all of them as a group, then for 3 of them, then for 13 of them, in each case for 3 formats times 5 samples, and that you also included two-tailed results in addition to one-tailed results.

It gives a total of [(16 x 5) + (16 x 3) + (1 +1 + 1) x 3 x 5] x 2 = 346 possible p-values, out of which you got 12 significant ones. However, out of 346 p-values, we should expect in average 346 / 20 = 17.3 of them to be significant by chance, that is false positives !

In reality, it is more complicated, because the 346 p values are not independant at all. The values for the 13 listeners group are close to the ones for the 16-listeners group. The values by format for the 3-listeners group are also conditionned by the fact that this group was selected according to their individual p-values for all formats etc.

But it is right that results by format, genre, and listeners should be evaluated independantly. The trick is to use a suited analysis that takes into account all these variables at once, but is not prone to false positive picking.

I don't know what method should be used in this case, but i'm sure that some forum members, more knoledgeable than me in statistics, can help.


Edit : correction of calculus.

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krabapple
post Jul 19 2010, 14:08
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QUOTE (Arnold B. Krueger @ Jul 19 2010, 06:48) *
QUOTE (amandinepras @ Jul 18 2010, 19:52) *
We used the Fireface DAC as it was the only one that allowed us to switch sample rate with a reasonable delay for the test (less than 1sec.). I wish we could use a better one. However, the Fireface is still pretty good compared to most playback systems people use in their house.


It's not clear which Fireface DAC that you used



Arny, it's named right there in my long summary post of the article, earlier in this thread:


"RME Fireface 800 DAC"


I also named the downsampling software (which someone else asked about)


amandine:
"there is no proof that upsampling doesn't introduce artifacts". There is reasonable inference to be made that upsampling can be *audibly transparent* , if not measurably perfect, based on what it does, how it's implemented, and how hearing works. What is known about those things does not go out the window just because we have no rigorous published test of that exact proposition to prove it. There's lots of more or less unlikely things for which there is no 'proof' in that sense, including the famous teacup orbiting the sun.


The burden on groups like yours is to demonstrate that reality confounds such reasonable models (and to replace it with a better reasonable model). My main issue with your paper is that your strongest evidence appears to be a statistically unlikely pattern of *incorrect* answers achieved by four subjects, on certain test signals only. Also troubled by the weird, inconsistent pattern of results or, e.g., 44.1 native vs 44.1 downsampled. And finally, it's not clear to me how switching was actually done. I get that it was via the Fireface, but did someone manually switch back and forth between SRs, or was this done via software? (The main effect I'd expect of slow switching times would be to *decrease* sensitivity to difference, so in itself this doesn't case doubt on the 'positive' results achieved. Too, one has to be very, very, very careful of subtly audible switching artifacts correlated with A and B, though again I would expect this to result in more *true positives* than you saw.)

As you say, clearly more work is in order. Your idea (in disucssion section) of focusing more on reverb tails in interesting but I would expect that to make manifest differences in *bit depth*, more than SR.

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Arnold B. Kruege...
post Jul 19 2010, 14:33
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QUOTE (googlebot @ Jul 19 2010, 07:19) *
QUOTE (Arnold B. Krueger @ Jul 19 2010, 12:48) *
Link to Fireface 800 technical test

These results don't look particularly impressive to me. They appear to be to be representitive of hi rez digital players at the lower end of the market.


Lets not criticize where no critic is due.

In any other context, you would have (rightfully) smashed TOS8, ABX, and your contradicting experience into anyones face, who had claimed that a DAC with the capabilities of the Fireface would not be sufficient for testing and comparing real world music.


Talking with hi-rez proponents is difficult because for openers, they flaunt TOS8. Their work inherently critizes the idea that conventional measures and criteria are sufficient. They must disresepct the work of the careful experimenters that have gone before them.

When dealing with them I am sometimes motivated for pity for the technological equivalent of dead horses. ;-)

In short, its hard to deal with hi rez proponents on the grounds of science and reason as we understand them.

So, searching about for some common ground, I noticed the claim that the performance of the Fireface 800 (which I have been cricitized for not discerning even though the evidence I presented about it was 100% on target) was characteristic of hi resolultion music players. My point was that it isn't. Check some relevant Stereophile test reports and see what I mean.


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googlebot
post Jul 19 2010, 16:28
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  1. The study shows at least the intent of objectivity and one of the authors even registered here to face up to possible criticism. So lets not treat them as nuts just for publishing something that could be read as proposing high rez delivery. I also believe, that the study, as it is, might have some statistical flaws. But lets give the authors credit for being here to discuss this.
  2. You find it hard to argue with high-rez nuts on the grounds of science and reason, still you criticize the study's authors for not playing by the nuts' rules and use a nut-compatible audio interface. That could be a valid hint if the study had shown clear signs of indistinguishability and was under heavy flax by high-rez nuts. But the tone of this discussion, so far, is rather why and if we have got significant positive results. So why not follow that path first? I don't see how criticizing the use of the Fireface helps here.


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Arnold B. Kruege...
post Jul 19 2010, 19:11
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QUOTE (googlebot @ Jul 19 2010, 11:28) *
[lI don't see how criticizing the use of the Fireface helps here.


Doesn't anybody else see a problem with using an audio interface with A-weighted SNR of only 89 dB in an attempt to prove that the CD format with *unweighted* SNR of more like 96 dB sounds worse?

If I was designing a test that designed to show the inadequacies of the CD format, I would think that the test system should have a SNR that is much better (at least 10 dB better) than that of the CD format. Otherwise, the audio interface in the test system is the weakest link, not the so-called "Low rez" CD format.

There are reasons why I own audio interfaces with 110 and 116 dB SNR...


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Juha
post Jul 19 2010, 19:25
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QUOTE
Doesn't anybody else see a problem with using an audio interface with A-weighted SNR of only 89 dB in an attempt to prove that the CD format with *unweighted* SNR of more like 96 dB sounds worse?


What makes you trust those RMAA results for FF800 you linked are valid? laugh.gif


Juha
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googlebot
post Jul 19 2010, 20:22
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The manufacturer's specs for the FF 800. Unweighted SNR is claimed to be <-100 dB, might be lower in 44.1 kHz mode, though.

QUOTE (Arnold B. Krueger @ Jul 19 2010, 20:11) *
There are reasons why I own audio interfaces with 110 and 116 dB SNR...


Do you expect even more significant differences between high rez and low rez with better gear? Do you expect that the music material used had anywhere close to -100 dB SNR? No & no. I smell much more desire for pissing match than reason, let alone any contribution to clear up why there might be a reported difference against all orthodox knowledge about the thresholds of human hearing.

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Soap
post Jul 19 2010, 20:24
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QUOTE (Arnold B. Krueger @ Jul 19 2010, 14:11) *
QUOTE (googlebot @ Jul 19 2010, 11:28) *
[lI don't see how criticizing the use of the Fireface helps here.


Doesn't anybody else see a problem with using an audio interface with A-weighted SNR of only 89 dB in an attempt to prove that the CD format with *unweighted* SNR of more like 96 dB sounds worse?

If I was designing a test that designed to show the inadequacies of the CD format, I would think that the test system should have a SNR that is much better (at least 10 dB better) than that of the CD format. Otherwise, the audio interface in the test system is the weakest link, not the so-called "Low rez" CD format.

There are reasons why I own audio interfaces with 110 and 116 dB SNR...


I was under the impression this was about 44.1 vs 88.2, not 16 bit vs 24 bit.


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Arnold B. Kruege...
post Jul 19 2010, 21:26
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QUOTE (googlebot @ Jul 19 2010, 15:22) *
The manufacturer's specs for the FF 800. Unweighted SNR is claimed to be <-100 dB, might be lower in 44.1 kHz mode, though.

QUOTE (Arnold B. Krueger @ Jul 19 2010, 20:11) *
There are reasons why I own audio interfaces with 110 and 116 dB SNR...


Do you expect even more significant differences between high rez and low rez with better gear?


In terms of measured differences, I would expect more significant differences between high rez and low rez with better gear. I would hope that my views on the adequacy of 44/16 as a distribution format for listening to music are well known. That means that I don't expect any audible differences to be found if the 44/16 stuff is up to snuff.


QUOTE
Do you expect that the music material used had anywhere close to -100 dB SNR?


Of course not! To get a reliable outcome not only does the music have to have a > 100 dB SNR, so does the listening room and the entire reproduction chain. Then there are the problems of the ear itself.

There are commercial recordings kicking around that have dynamic range on the order of 85 dB, but that is it. To listen to them in a normal listening room with say a 45 dB SPL noise level the peaks would have to be about 130 dB SPL which is more than enough to cause a signficiant threshold shift in the listener's ears.

QUOTE
No & no.


Of course, but we're dealing with people who don't know that they are on mission impossible and don't believe us when we suggest that they are. The only way they are going to change their minds is if they prove it to themselves. As long as their experiement has holes big enough for me to drive a truck through, their hope can always spring eternal. I'm just trying to help them do the cleanest experiment they can from their own viewpoint.

QUOTE
I smell much more desire for pissing match than reason, let alone any contribution to clear up why there might be a reported difference against all orthodox knowledge about the thresholds of human hearing.


I think you need to step back a second and look at the facts that are before you. When I was doing experiments like these I did exactly what I'm trying to suggest to these folks. I used audio interfaces that had 105, 110 and 116 dB dynamic range. Do I need to post the S/N of my LynxTWO to prove that I still have it? If I'm the person you seem to want to libelling me as being, why did I do that (at my own personal expense)? Hint: it had nothing to do with piss. It was all about a search for truth.

IMO part of a good clean job of studying a situation may include overkilling it from a quality standpoint, even when "You know better".

When I did these experiments back about 9-10 years back, I didn't know all of the reasons why it couldn't work that I know today. Once I had the results before me, I gathered relevant facts to confirm or deny the result that I had before me. Then theory matched results and I could confidently move on.

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post Jul 19 2010, 23:07
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QUOTE (Arnold B. Krueger @ Jul 19 2010, 14:11) *
Doesn't anybody else see a problem with using an audio interface with A-weighted SNR of only 89 dB in an attempt to prove that the CD format with *unweighted* SNR of more like 96 dB sounds worse?

I still don't see this apparent attack justified.

How does the interface's A-weighted SNR of only 89 dB factor into the ability to distinguish 44.1 from 88.2 material?

I thought this was not a 16 v 24 test but a 44.1 v 88.2 test.



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Cavaille
post Jul 20 2010, 08:40
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Forgive me to interrupt this discussion about the Fireface 800 (which seems to be a decent interface - or does it contain another matter?!?) but I have a question.

Arnie, you suggested to improve this test by upsampling the downsampled material to the samplerate the original had. I very much see the logic in your suggestion. However, may it be that brick wall filtering could introduce audible obstacles into the signal that are unwanted? Im referring to the thread "Audibility of 20kHz brick wall filtering". So far (only three people have participated, including me) it seems that brick wall filtering may be audible. Further tests by several people are required however. And it is my impression that both downsampling & upsampling use brick wall filtering to avoid aliasing artifacts for downsampling and imaging products for upsampling. Is that assumption correct?

So, if Im using brick wall filtering two times, wouldnt that be even more audible? Or am I getting this wrong?


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