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  #16  
Old 11-02-17, 02:27 AM
Jim Audiomisc Jim Audiomisc is offline
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FWIW I used TPDF in my demo as I regard that as the basic standard.

And AIUI the reason 'brick wall' (i.e sinc-based) filtering was widely adopted from the start is that this should then simply 'pass though' whatever the effective impulse response of the system used to generate the *recording* may have. i.e. it was assumed its the job of the recording engineers, etc, to decide what they wanted you to get.

All that said, I used a first-get Marantz player for about a decade quite happily. Although I *did* add another analogue low-pass filter after it. This was an old 'Toko' passive filter design they made for Yamaha's 'pilot tone nulling' FM tuners. These gave a flat response up to about 18kHz and then rolled smoothly as they didn't need to kill the 19k pilot. Yamaha's auto-nulling did that. Worked quite nicely for the CD player. :-)
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  #17  
Old 11-02-17, 02:29 AM
Jim Audiomisc Jim Audiomisc is offline
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I did experiment with using SoX for the requantization. However I found that it refused to let me employ any of the named shapings it lists. This was for 192k/24 -> 192k/16. My conclusion was that the shapings it offers are directed at producing 44.1k/48k output so only have coefficients for that. Did I miss something?
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  #18  
Old 11-02-17, 02:38 AM
Werner Werner is offline
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Quote:
Originally Posted by darrenyeats View Post
A reconstruction filter rolling off 19-22kHz sounds better to me at the top end - for some recordings anyway - and I don't feel I'm missing anything. In fact the opposite. I don't know whether this is:
1. Simply the amps/speakers being asked to produce less energy >19kHz.
2. A gradual filter roll off. Does linear phase mean this should not matter?
Quote:
I assume brick wall filters are done because they are easy/cheap/low processing power though.
The ideal reconstruction filter, sinc(x), is infinitely steep. It stands to reason that the industry attempts to approach this for digital replay. (But doing so, it ignores the ADC side of things, but that is a different story). So it is to be expected that most commercial CD filters, at least before the often entirely misunderstood apodising craze started, are quite steep and cut at 22kHz. Making them steep adds cost. Making them cut at exactly Fs/2 halves the number of coefficients, reducing cost again. Thus symmetrical linear phase half-band FIR. Since steepness is not infinite, imaging occurs.

This would be perfectly innocuous if the recording side was done properly, meaning with zero content at Fs/2. In such case the DAC side would not ring (since zero content at Fs/2), and would generate a minimum of images, since the ADC side already ensured that there is not much near Fs/2.

But that is not how industry works. Seeing those nice and economic linear phase half-band FIR filters in DACs, they decided to adopt the same filter style for in-ADC downsampling (and, broadly, also in downsampling software, which, until recently, was abysmally bad, see ProTools, Pyramix, Merging, ... at src.infinitewave.ca). So the ADCs also got steep half-band filters. These ensure a couple of bad things:
-aliasing happens, infecting the 20-22kHz part of the recording
-strong ADC filter ringing is inserted at 22kHz
-DAC-side ringing is triggered
-DAC-side imaging is triggered

In short: industry practice guarantees that the 20-24kHz range of CD is buggered during replay. Can't hear it directly, but metal domes will ring, and systems with IMD issues may show this up.


And yet, the recipe is simple, provided we drop that old fetish of needing flatness to beyond 20kHz.

Just somewhere in the chain, ideally at the ADC side, start rolling off at 18kHz, and reach zero, or at least a suitably low level (most music hasn't got that much of high treble anyway), at Fs/2. That gives the filter a 4kHz-wide transition band. 4kHz is also, give or take, the width of the highest critical band in the ears of healthy young people. This then means that the filter's ringing is of the same order as the innate temporal acuity of that highest critical band, i.e. good enough. So in one step you ensure that:
-DAC-side ringing is not triggered
-DAC-side imaging is not triggered
-any ringing is inaudible, even for those with fresh ears
-the downstream system is only fed with the music, nothing else.

Quote:
they are easy/cheap/low processing power. But filter shape aside, fair to say a lot digital filtering falls significantly short of current SOTA (which I believe is Saracon), and
iZotope is the reference (*). Has been for a long time.


(* When configured properly. Last year it was very fashionable at CA to dream up iZotope settings and validate them by ear. A lot of garbage was generated, and people loved it. Of course, the tiniest settings differences were always day-and-night audible.)
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  #19  
Old 11-02-17, 02:43 AM
Tony L Tony L is offline
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Quote:
Originally Posted by Jim Audiomisc View Post
i.e. it was assumed its the job of the recording engineers, etc, to decide what they wanted you to get.
Giggles. IME any such recording system is bought at great cost, unpacked, hooked up, the "engineer" spends best part of a day swearing at some box called a 'word clock' and tries to figure out why Cubase is exactly one hour out of SMPTE sync with the f***ing ADAT and when the right combination if swear words are found to rectify the situation it is never spoken of again. I'd be utterly amazed if you could find many/any studio sound engineers who understand this stuff down to the bits, bytes, noise shaping, filters etc level. Basically if most of what you are monitoring sticks to the tape you are done!

PS Obviously you do all this in Logic Pro X or whatever on a MacBook these days!
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  #20  
Old 11-02-17, 03:02 AM
davidsrsb davidsrsb is offline
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Quote:
Originally Posted by Werner View Post
Just somewhere in the chain, ideally at the ADC side, start rolling off at 18kHz, and reach zero, or at least a suitably low level (most music hasn't got that much of high treble anyway), at Fs/2. That gives the filter a 4kHz-wide transition band. 4kHz is also, give or take, the width of the highest critical band in the ears of healthy young people. This then means that the filter's ringing is of the same order as the innate temporal acuity of that highest critical band, i.e. good enough
So given that existing recordings were not filtered properly, a post processing stage of a 18kHz 3dB point, 4th order or more low pass should reduce these artifacts?
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  #21  
Old 11-02-17, 03:12 AM
Werner Werner is offline
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Quote:
Originally Posted by davidsrsb View Post
So given that existing recordings were not filtered properly, a post processing stage of a 18kHz 3dB point, 4th order or more low pass should reduce these artifacts?
Ironically pre-oversampling era PCM1610/1630 recordings behave rather nicely in the upper treble, from an aliasing PoV.

Half-band recordings, on the other hand, should be replayed with an oversampling filter as I described. I don't think there is any DAC or CD player manufacturer doing this. They are all still set on printing "frequency response 20Hz-20kHz +/-0.1dB" in their blurb.

But we are deviating ...


Earlier question: why no cascaded noise shaping?

Well, each noise shaper is built on assumptions about audibility of the resulting noise/distortion. If you mix noise shapers, you start mixing assumptions, and the outcome may well be a mess.
If there is any chance that a noise shaping process is living downstream the signal path, then better apply TPDF at your own stage.
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  #22  
Old 11-02-17, 03:39 AM
darrenyeats darrenyeats is offline
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Thanks Werner, to me that's new and interesting information about the why. In terms of the what, it seems I'm on the right track with a transition band from 19-22kHz (and I will now try 18-22kHz)!
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Last edited by darrenyeats; 11-02-17 at 09:22 AM.
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  #23  
Old 11-02-17, 06:30 AM
Jim Audiomisc Jim Audiomisc is offline
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Quote:
Originally Posted by Tony L View Post
Giggles. IME any such recording system is bought at great cost, unpacked, hooked up, the "engineer" spends best part of a day swearing at some box called a 'word clock' and tries to figure out why Cubase is exactly one hour out of SMPTE sync with the f***ing ADAT and when the right combination if swear words are found to rectify the situation it is never spoken of again. I'd be utterly amazed if you could find many/any studio sound engineers who understand this stuff down to the bits, bytes, noise shaping, filters etc level. Basically if most of what you are monitoring sticks to the tape you are done!
Alas, yes. I fear the Philips/Sony people did totally misjudge the ability of some in the 'music biz' to use boxes without any clue or care wrt what they were *actually* doing. Hence all the recordings that are clipped or have other problems like badly scaled levels causing patterns in the sample distribution and lousing up the results.

All made worse, I guess, by the tendency to bring in 'gurus' who can 'make it sell' ... by fouling it up. Loudness, anyone? :-/
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  #24  
Old 11-02-17, 06:37 AM
Jim Audiomisc Jim Audiomisc is offline
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Quote:
Originally Posted by Werner View Post
And yet, the recipe is simple, provided we drop that old fetish of needing flatness to beyond 20kHz.

Just somewhere in the chain, ideally at the ADC side, start rolling off at 18kHz, and reach zero, or at least a suitably low level
Agreed. In part I guess this is why my old trick of using a decent analogue filter after reconstruction that removed cut away the stuff about about 18k sounded better quite often.

I wonder if one root of the problem was/is that many engineers who have even learned how the sinc shape is derived from IT realise that to be formally 'correct' it has to cover *all* the samples in the recording and have perfect resolution. So any real filter has to be an approximation. Hence the wisdom of keeping the danger zone clean of anything that might cause a problem.

As on some earlier occasions, I regret that you can't now easily buy those old Toko filters. They were quite good analog designs. All killed off, I guess by digital filtering.
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  #25  
Old 11-02-17, 08:02 AM
davidsrsb davidsrsb is offline
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At any time in the last 30 years, there have always been the odd CD player out there with a ~18 kHz low pass. These have usually been said to sound "more analogue" but "slightly dull".
I think few reviewers can hear anything like 18 kHz, so this is more likely down to the absence of dome tweeter break up hash triggered by artifacts
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  #26  
Old 12-02-17, 12:40 PM
Richard Kimber Richard Kimber is offline
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So, if one has the choice (e.g. HQPlayer) how does one choose which noise shaping to use? (Apart from just trying each one and forming a preference.) I.e. are there good reasons to prefer one approach rather than another?

- Richard.
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  #27  
Old 12-02-17, 01:02 PM
darrenyeats darrenyeats is offline
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This is discussed in post 15 and last part of post 21.
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  #28  
Old 13-02-17, 08:25 AM
Richard Kimber Richard Kimber is offline
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Originally Posted by darrenyeats View Post
This is discussed in post 15 and last part of post 21.
Not really my idea of a discussion of the question I asked.
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  #29  
Old 13-02-17, 08:43 AM
adamdea adamdea is offline
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Quote:
Originally Posted by Richard Kimber View Post
Not really my idea of a discussion of the question I asked.
No. What to do about noise shaping dither at 16/44 is quite different from noise shaping with a single bit at MHz sample rates.
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  #30  
Old 13-02-17, 09:07 AM
Jim Audiomisc Jim Audiomisc is offline
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Quote:
Originally Posted by Richard Kimber View Post
So, if one has the choice (e.g. HQPlayer) how does one choose which noise shaping to use? (Apart from just trying each one and forming a preference.) I.e. are there good reasons to prefer one approach rather than another?

- Richard.
TBH I'm not clear what question you really have in mind. The choice of a reconstruction filter is a different matter to the choice of Noise Shaping. Although a given filter may well employ shaping.

For the end-user the simplest approach is to just 'play out the samples' if your DAC can accept them. If you're worried about 'ringing' or 'artifacts near the top of the band' then apply a filter. However if you just want to slope off the top off the band or similar, the filter shape is probably the main issue.
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