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Old 22nd July 2010, 01:11 AM   #5021
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Quote:
Originally Posted by john curl View Post
I do not look at nameplates.
Except for the invisible nameplate that says 'digital'. Then its 'condemnation without examination' for sure.
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Old 22nd July 2010, 01:29 AM   #5022
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The Boston test is a problem. Although i doupt that the speaker setup was state of the art
( free of horizontal early reflexions that disturb imaging, see my AES paper with Malkolm Hawksford and Bernd Theis) no one has proven so far that this test did not tell the truth.
When digital processing is carried down to the speaker results from 16/44 digital can be excellent. We build a digital speaker in Essex in the early 90th that sounded mesmerising,
still my speaker system at home sounds even better to my ears. No digital processing is used in my reference spaekers but they are phase coherent. Yes i know, there are a lot of definitions what phase coherent means. I mean that the hilbert transform of the transfer function of my spaekers is also the measured phase response. In Europe we call that a causal system.
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Old 22nd July 2010, 01:32 AM   #5023
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Being analog they have no pre- and very little postringing.
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Old 22nd July 2010, 01:55 AM   #5024
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Why should they be considered 'superior' IF they usually do not reflect what people hear consistently on their own. Perhaps it is the ABX test that is at fault. I think that is so.
It may indeed be a problem with the ABX test. Why do you think its that rather than people's perceptions being influenced by what they think they know about the equipment (expectation effect/ confirmation bias) ?
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Old 22nd July 2010, 02:50 AM   #5025
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Terry, you need to explain something here.

"Down sample to 44.1 / 24 using a) Tony Faulkners averaging program (no
LPF) and b) a high quality SRC program (typically with with brickwall
filtering)."

In this context downsampling, to me at least, is a sample rate conversion and the brickwall filter defeats the point of the TF averaging. I'm confused. I read the Stereophile article (in part) more Wadia-esque stuff about deliberately letting some aliasing(imaging) get in to avoid the dreaded Gibb's effect. Please show me some real music waveforms that exhibit significant pre-ringing. Again we are criticised because "music is not sine/square waves" but are shown impulse and square wave responses as examples of how the mathematically correct way of doing things "looks like it must sound bad". Greiner did careful DBT's that showed brickwall filters are not audible. He is a good engineer and his tests were very careful and peer reviewed.
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Old 22nd July 2010, 03:40 AM   #5026
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Quote:
Originally Posted by scott wurcer View Post
Terry, you need to explain something here.

"Down sample to 44.1 / 24 using a) Tony Faulkners averaging program (no
LPF) and b) a high quality SRC program (typically with with brickwall
filtering)."

In this context downsampling, to me at least, is a sample rate conversion and the brickwall filter defeats the point of the TF averaging. I'm confused. I read the Stereophile article (in part) more Wadia-esque stuff about deliberately letting some aliasing(imaging) get in to avoid the dreaded Gibb's effect. Please show me some real music waveforms that exhibit significant pre-ringing. Again we are criticised because "music is not sine/square waves" but are shown impulse and square wave responses as examples of how the mathematically correct way of doing things "looks like it must sound bad". Greiner did careful DBT's that showed brickwall filters are not audible. He is a good engineer and his tests were very careful and peer reviewed.
Scott,

All points noted and well understood.

If people listen to in isolation the various mechanisms to get from 24/176.4
to 16/44.1 we might have more informed opinions.

I preferred the TF downsampling to a HQ brickwall SRC but it was pretty
subtle. The material was acoustic / non complex and I am also completely
comfortable that it is very likely a euphonic effect.

We've been doing it for years with tape \_/

T
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Old 22nd July 2010, 03:51 AM   #5027
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Originally Posted by Terry Demol View Post
If people listen to in isolation the various mechanisms to get from 24/176.4
to 16/44.1 we might have more informed opinions.
I'm curious to know if you (and others) hear the same things with material recorded directly in 24/176.4 i.e. not converted from DSD. There might be something going on related to the out of band noise inherent in DSD.
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Old 22nd July 2010, 04:00 AM   #5028
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Jan & all others - let's take the ABX debate off line...

tnx.

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Old 22nd July 2010, 04:26 AM   #5029
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Quote:
Originally Posted by abraxalito View Post
Its obscure to me that higher order filters necessarily provide faster settling. As far as I can see, the settling time isn't so much a function of the filter order as the Q - a higher order bessel filter will settle quicker than a somewhat lower order elliptic for example.



I don't consider a level error (linear) of 1% to be particularly worrisome. Do you? The faster the settling, the smaller this level error. 5 time constants will give an error -43dB down.



Sorry, I'm not following here. Both methods?
For the same type of filter the higher the order the greater the out of band attenuation and the faster you should get to your design goal.

Yes 5 time constants of error on a step size of 1/2 full scale is significant.

In a ratiometric converter the maximum change is a single bit and both methods (linear filter and inverse exponent) are about equal. In other D/A methods you can have a change from one output period to the next of up to full scale of the converter. So if your settling error is -44db that is probably not very good.

Last edited by simon7000; 22nd July 2010 at 04:29 AM.
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Old 22nd July 2010, 04:35 AM   #5030
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Quote:
Originally Posted by simon7000 View Post
For the same type of filter the higher the order the greater the out of band attenuation and the faster you should get to your design goal.
Sure, but that's another design goal, related to aliasing, not settling time. Or have I missed some relationship between the two?

Quote:
Yes 5 time constants of error on a step size of 1/2 full scale is significant.
I could understand if this was a non-linear error, but its entirely linear - just results in a level change. Stereo pots (just to choose a random example) introduce far greater level errors - they'd need to be jolly good to maintain their tracking error below -43dB.

Quote:
In a ratiometric converter the maximum change is a single bit and both methods (linear filter and inverse exponent) are about equal. In other D/A methods you can have a change from one output period to the next of up to full scale of the converter. So if your settling error is -44db that is probably not very good.
I'm still confused. How are linear filters and inverse exponents different? To me an inverse exponentially settling filter is a linear filter. If you think its non-linear, please explain how this comes about.
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