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Hi Charles, I'm not knowledgeable about electronics at all. I read your white paper on Minimum Phase and thought I had a basic understanding of what your Minimum Phase filters were doing. Here is a quote from the white paper describing the effect of an apodizing filter,
"Furthermore all pre-ringing from the recording process has been
filtered out, and the new playback filter only has post-ringing."My understanding of the white paper is that your MP filter builds on the apodizing filter's removal of the pre-ringing by minimizinging the post-ringing.
Then I read the following in your post below, "It is this apodizing filter that is *supposed* to filter out problems created by the ADC. But I can't say that the thing really works. It doesn't sound bad, in fact it sounds pretty good. But it doesn't sound as good to me as our "Listen" filter that *doesn't* filter out the alleged ringing that the ADC puts on the disc."
So I'm confused, can you briefly explain what your Listen filter is removing? Thanks, Tom
Edits: 09/14/09Follow Ups:
Where to start ...
First this: the term 'apodising filter' is not really defined. It was adopted some 5 years ago from astronomics by Peter Craven who needed a name for a anti-aliasing (ADC) or anti-imaging (DAC) filter that in order to 1) reach full attenuation before Fs/2 and 2) still have a short impulse response, 3) sacrifices a significant amount of the pass band (i.e. has a slow roll-off). These first filters where defined for 96kHz and 192kHz, and started rolling off beyond 20kHz and 40kHz respectively.
Now with CD there is not much of the pass band available for sacrifice (unless you want -3dB at 15kHz or so), but the apodising label stuck and now also denotes any filter that is flat out to 19kHz or so and reaches full attenuation before Fs/2=22kHz.
More later ...
bring bac k dynamic range
Digital anti-alias (ADC) and anti-imaging (DAC) filters can broadly be characterised by1) cut-off point
2) slope
3) phase
-----
The cut-off frequency can be:
a) below Fs/2: e.g. at 20kHz instead of 22.05kHz, this is sometimes done
b) at Fs/2: typically implemented as 'half-band' filters since these require 50% of the resources of type a). Since there cannot be full attenuation at Fs/2, some aliasing or imaging occurs. The amount of aliasing/imaging depends on the slope: little with steep, much with slow.
c) above Fs/2: I don't think this has any application in quality audio.
The filter slope can be:a) fast, steep: as a result, there is much pre- and/or post-ringing in the filter's impulse response.
b) slow: there is little pre- and/or post-ringing.
The filter phase can be:a) linear phase: no phase shift, there is pre- and post-ringing.
b) minimum phase: there is phase shift, and the phase-frequency function equals the Hilbert transform of the amplitude-frequency function (as with analogue filters). There is only post-ringing.
c) a mixture of both. There is pre- and post-ringing, although less pre- than post-.
Still more later ...
Edits: 09/15/09
Let's have a first look at the replay side:
The theoretically ideal DAC anti-imaging (aka reconstruction) filter is defined as the Sinc(t) function. It is characterised as:
-cut-off: at Fs/2
-slope: infinitely fast => a lot of ringing, no imaging
-phase: linear => pre- and post-ringing
It has a lot of pre- and post-ringing, but this can be proven to be of zero (0) influence, provided the source signal was correctly band-limited prior to sampling. This filter can't be realised, but it can be arbitrarily approximated, especially in software-based oversamplers.
Most commercial DACs and replay SRCs (HW and SW) are a decent approximation of the above:
-cut-off: at Fs/2 (half-band for economy) => a bit of imaging
-slope: fast => a lot of ringing at Fs/2 (22kHz)
-phase: linear => pre- and post-ringing
Now to the recording side:
Contrary to the replay side there exists no ideal model of an anti-aliasing filter. The sampling theorem demands a correctly band-limited input signal but gives no clues as to how to do this.
Consequently any anti-aliasing filter for CD production has to be a compromise. Disregarding analogue AA filters, there are two commonly-used digital filter types:
The one most-often used in single-chip ADCs:
-cut-off: Fs/2 (half-band, again for economy) => some aliasing
-slope: fast => lots of ringing
-phase: linear => pre- and post-ringing
This type of filter embeds pre- and post-ringing at 22.05kHz in the recorded signal. Moreover, as it allows some aliasing and as its attenuation is only 6dB at Fs/2, it excites ringing in the playback DAC too (since this sort of AA filter mildly violates the sampling theorem).
Sometimes used in decent software SRCs:
-cut-off: somewhat below Fs/2, say 20kHz or 21kHz
-slope: fast => lots of ringing, but combined with the lower cut-off less or even no aliasing
-phase: linear.
This type of filter still embeds pre- and post-ringing, but it does not excite the DAC's ringing.
Time for a summary:
1) almost all commercial ADCs or production SRCs embed pre- and post-ringing at 20-22kHz into the recorded signal.
2) many commercial ADCs allow some aliasing. This aliasing triggers the otherwise-innocuous pre- and post-ringing of Sinc(t)-based DACs
3) almost all commercial DACs are half-band Sinc(t) approximations.
At this point the presence of pre-ringing is perceived as a bad thing (although the audibility of this has not been proven!!!), and some remedy is wanted.
More later ...
bring bac k dynamic range
Back to DACs and their anti-imaging reconstruction filters.
In the late 80s first Wadia and then many others opted for slower filters:
-cut-off: at or below Fs/2
-slope: slow, audible treble losses below 22kHz, significant imaging above 22kHz
-phase: linear, a little bit of pre- and post-ringing
Viewed on their own and through marketeers' eyes these filters looked like a cool idea, as the impulse response had significantly less ringing (compared to Sinc(t)), while the price to pay was a bit of treble loss (who hears that?) and a bit of imaging (harmless for a decent amplifier).
Viewed in combination with the recording side of course the ADC's embedded pre- and post-ringing rears its head again, destroying the marketeer's argument. Still, many listeners preferred this type of filter
(perhaps because they are computationally much less complex than regular Sinc-like filters?)
A limit case of the above is the Non-Oversampling Filterless DAC:
-cut-off: none
-slope: infinitely slow
-phase: linear
Such a DAC's impulse response is a perfect needle with zero ringing. Impressive to the layman, an utter disaster under the sampling theorem. Needless to say that the system response is now entirely dictated by the ADC's ringing and the DAC's copious amount of imaging products. Really the worst of two worlds.
More later ...
bring bac k dynamic range
So now we have a number of fancy DAC filter styles, but none of these do something about the ADC's embedded pre-ringing.
Enter then Meridian, who brought Craven's apodising ideas to CD and who also were the first to get rid of the notion that only linear phase filters should be used.
The Meridian 808.2-whatever replay filter is:
-cut-off: below 20kHz ('apodising')
-slope: fast => lots of ringing, no imaging
-phase: minimum => no pre-ringing, lots of post-ringing
The crux of this filter is that it literally under-cuts most ADC filters, and thus attenuates the embedded ADC ringing. It imposes its own ringing at 20kHz, but since this is a minimum phase filter this is only post-ringing, not pre-. Found in Meridian, Ayre MP-Measure, and some new products.
And then there is Ayre's latest MP-Listen filter:
-cut-off: Fs/2, half-band (as per Ayre's paper)
-slope: slow => little ringing, quite some imaging
-phase: minimum => no pre-ringing
Since this filter only attenuates 6dB at 22kHz it does not under-cut the ADC's filter, and thus the ADC's embedded pre-ringing largely remains in the signal. An aliasing ADC triggers the DAC's ringing, but this is only a little bit of post-ringing. This type of filter should be seen as the minimum-phase version of the older Wadia type.
bring bac k dynamic range
I am not sure why, though, the MP Listen filter is such a big step beyond filterless NOS designs since it only provides a modest amount of anti-aliasing (eyeballing the graph in the paper looks like between 3-12db of attenuation from 20-28kHz). Whereas the filterless design does not provide any obviously, it is still only an incremental advantage.
Edits: 09/16/09
> > I am not sure why, though, the MP Listen filter is such a big step beyond filterless NOS designs < <
Me either.
I think it boils down to the fact that Redbook is such a compromised format. So the best way to get good sound is with a filter that is a compromise between two extremes.
At one end you have non-oversampling (NOS) filterless DACs. At the other extreme you have a brickwall filter.
We tried all kinds of different things. We started with an NOS filterless design to see what all the fuss was about. It made a pretty dramatic improvement overall, but it also had some sonic problems. But we were actually changing two variables at once in that experiment -- one was going from 32x to 1x. The other was bypassing the digital filter inside the DAC chip.
Next we tried our own 4x external filter but still bypassing the digital filter in the DAC chip. That was even better! So a big problem was the filter inside the DAC chip. I don't know if this is because the filter was a poor one or if just the fact that you put a high speed digital circuit right next to the delicate analog circuitry introduced some RFI into the world. But it didn't really matter, because we could just build whatever filter we wanted outside the DAC chip in an FPGA.
We tried dozens of different filters We even made an apodizing non-oversampling filter with a notch at 22.05 kHz! Seemed like a good idea -- the benefits of NOS the benefits of an apodizing filter all in one. But it really didn't sound noticeably different from the straight NOS filter. (Which is another reason why I think that the hypothesized ringing on the recording that the apodizing filter is supposed to eliminate is a good theory, but not really what is happening.)
After a while, certain patterns started to emerge. Minimum phase consistently sounded better than linear phase filters. Going from 1x to 4x to 8x to 16x oversampling made a clear improvement each time. Gentle filters with less ringing sounded better than sharp filters with a lot of ringing.
The interesting one was choosing the coefficients. There are dozens of algorithms that have been described:
* Rectangular
* Hanning (Hann)
* Hamming
* Triangular
* Blackman
* Exact Blackman
* 3 Term Cosine
* 3 Term Cosine with continuous 3rd Derivative
* Minimum 3 Term Cosine
* 4 Term Cosine
* 4 Term Cosine with continuous 5th Derivative
* Minimum 4 Term Cosine
* Good 4 Term Blackman Harris
* Harris Flat Top
* Kaiser
* Dolph-Tschebyscheff
* Taylor
* Gaussian
are the ones offered in our filter design software. Each one made audible differences, with no apparent pattern. And for each filter there are dozens of different combinations of parameters -- passband frequency, stopband frequency, passband ripple, and stopband depth.
Like I said, we spent months to find the best sounding one. I don't think there is any other way to do it. Redbook is such a compromise. It barely works. It's right on the ragged edge of acceptability. When everything is done properly it can sound surprisingly good. And in the case of a compromised format, it would seem that a compromise in filter designs is the best solution.
to "perfect" Redbook (i.e. minimize the downside while maximizing its potential). I do think that these types of filters are destined to replace upsampling as the final destination for Redbook and I can't wait to hear one of these players.
> > I do think that these types of filters are destined to replace upsampling < <
"Upsampling" is a marketing buzzword. It doesn't mean anything in particular. Whenever we used it in our product literature, we always put quotes around it. Same thing for "DSD". It's not a technical term used by engineers. It's just something that the marketing folks thought sounded really cool.
Hi Charles,
you must be using Digital Filter Design software. That list is straight out of the marketing slide.
Did you try implementing the same (well at least similar) function with different number of taps? For example a sinc with 8 taps is fairly slow, but you can implement the same function with 512 taps (but its not a sinc anymore) I haven't done that experiment yet and wonder if there is an audible difference.
I came to the same conclusion, that the best approach is a tradeoff, high image attenuation sounds bad, but you have to have SOME attenuation otherwise it sounds bad in a different way. There is such a huge space in that in between area it would be nice if we had an objective function to optimize for, but unfortunately its pretty much try all kinds of different things and listen.
Hopefully over time we can come up with some good correlations and develop some objective functions. But maybe by then 44.1 will be dead and we won't need it!
BTW my current DAC in the listening room is NOS with an analog filter, this also sounds quite good, but it takes really good parts, so its a pretty expensive option. The digital filter is a lot cheaper and more flexible. I wanted to see if an analog filter sounded similar to a FIR with similar function and it does.
It turns out that you have to be very careful with the analog filter, most implementations will have ultrasonic resonances that are easily excited by the large amount of HF from the DAC chip. These resonances cause all kinds of sonic problems. I have a suspicion that these resonances have been the reason that NOS followed by analog filter has not been very successful.
John S.
> > you must be using Digital Filter Design software. < <
We use the package from Momentum Data Systems. With all the bells and whistles (necessary for minimum phase FIR filters) it is $2,000. For those who are interested there is a demo version that does everything except spit out the coefficients. You can have lots of fun with that.
> > Did you try implementing the same (well at least similar) function with different number of taps? < <
Basically there is a one-to-one correlation between the number of taps and the sharpness of the knee. Our "Measure" filter uses over 1000 taps, but our "Listen" filter only uses less than 300. The reason those numbers are so high is because we are doing 16x in one pass. When we used to do 4x externally and 8x inside the DAC chip, the external filter only needed 1/4 as many to achieve the same response.
"reason why I think that the hypothesized ringing on the recording that the apodizing filter is supposed to eliminate is a good theory, but not really what is happening."
There are broadly four types of recording-side anti-aliasing filters, and only one of them falls fully under the apodising story:
-minimum phase, with or without a bit of aliasing (mimick PCM1610/1630 here)
-linear phase, no aliasing
-linear phase, half-band (as most modern ADC chips)
The Meridian-style apo filter only attacks the fourth. What it does with the third depends entirely on whether the AA filter cut-off is above or below the DAC's. And in the case of the first two the apo filter's phase shift just adds to the AA filter's.
It might be enlightening to prepare a really well-done recording in
88.4kHz and then subsample it with the four AA classes.
These four source tracks can be used in the subjective evaluation of your 22^14^18^4 possible replay filter types ;-)
bring bac k dynamic range
"These four source tracks can be used in the subjective evaluation of your 22^14^18^4 possible replay filter types ;-)"
Subjective just doesn't cut the mustard on these forums - you should know that by now surely?
What is needed is a panel of listeners incapable of any emotional involvement or accepting any external influences whatsoever.
What do parking meter attendants do on their day off?
Best Regards,
Chris redmond.
"listeners incapable of any emotional involvement "
Do you know about the BBC's procedures for subjective evaluation?
They train the listeners on particular programme material until
their emotional response to that music/sound is zero.
bring bac k dynamic range
"Do you know about the BBC's procedures for subjective evaluation?
They train the listeners on particular programme material until
their emotional response to that music/sound is zero."
Are you talking about evaluation of sound quality or are you talking about auditions of musicians they will be employing? :-)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
"They train the listeners on particular programme material until
their emotional response to that music/sound is zero."
That's a coincidence, as nowadays I have exactly the same response to the majority of the BBC's television programme's after just one viewing!
Best Regards,
Chris redmond.
x
Regards,
Geoff
I'm confused too. What is your question?
Hi Charles, I got the impression from reading the white paper on MP that it eliminates pre-ringing and minimizes post-ringing, but in an excerpt from your post on 9/8 titled "Two Different Questions" you state, "our "Listen" filter that *doesn't* filter out the alleged ringing that the ADC puts on the disc."So, I own a CX-7eMP and think it sounds great but I've been telling people that it removes measureable, audible, unnatural distortion (pre- and post-ringing) but that doesn't appear to be the case from your post. So my question is, what does the MP filter remove? Is it ringing caused by playback or what?
I would appreciate just a simple answer. Werner has graciously given the long answer. Thanks Werner! I will study your posts. I am not trying to cause an argument I would just like to be able to explain to people why players with the MP filter sound better.
Tom
Edits: 09/15/09
First of all you keep talking about "the MP filter". There is no "the MP filter". There are *two* filters, both of which are minimum phase ("MP") types as we found that minimum phase filters sound better in controlled listening test than do linear phase filters.
One of the filters (labeled "Measure") has a sharp cutoff at 19.5 kHz and therefore gives flat frequency response in virtually the entire audio band. The cutoff is sharp enough that there is significant attenuation at 22.05 kHz. This is the frequency at which any ringing encoded on a disc would be at. So *if* there is ringing encoded on the disc, it will be virtually eliminated by using the "Measure" position.
The penalty for such a sharp rolloff is that the filter in the player will ring. Since it is a minimum phase filter, the ringing is all *after* the transient. Therefore using the "Measure" filter trades any pre- and post-ringing that *may* be present on the disc for purely post-ringing during actual playback of that disc.
The "Listen" filter is also a minimum phase filter. As shown in the white paper (and as Werner correctly notes), it only has -6 dB of attenuation at 22.05 kHz. So if there *is* any ringing encoded on the disc, it will only attenuate it mildly. The advantage of this filter is that it has a gentle rolloff and therefore contributes very little of its own ringing to the actual playback of the disc.
Since the "Listen" filter is nearly universally preferred over the "Measure" filter, I can only conclude that the Craven's hypothesis of ringing encoded on the disc is incorrect. If it were really there, then the "Measure" filter would sound better because it would get rid of it. But the "Listen" filter sounds better, so it must be more important for the playback filter not to ring.
However, the ultimate expression of "less ringing in the playback filter is better" is the filterless "non-oversampling" DAC. While this has its sonic charms, it doesn't sound as good as the "Listen" filter. The bottom line is like Werner said. Redbook CD is a compromise. There is no perfect solution, only tradeoffs. The "Listen" filter produces the best sounding tradeoff that we were able to create, and is demonstrably better than Craven's apodozing filter that we used in the "Measure" position.
Hi Charles.
I have tried using 24/96 files and even 24/176 files and resampled it down to 16/44 with Izotope RX using both linear phase sharp cutoff, and minimum phase. Accoording to http://src.infinitewave.ca/ Izotope is flat out the best resampler out there, and I know Barry Diament is also using this. (He prefers linear phase, sharp cutoff preset)
Well, I am using the QB-9 (only in listen, havent tried the measure position yet doing this comparison)
Anyways, the minimum phase src sounds clearly better than the linear phase src, and that is with a minimum phase DAC as QB-9 is.
I dont know if this is useful information or not, but it is kinda interesting since it shows that minimum phase might benefitable at both ADC and DAC. Also the reason why Barry prefers the linear phase SRC is because he has a linear phase DAC? What is you take on this?
Sorry, I don't know enough to make a knowledgeable comment. I don't know Barry Diament, I have never used Izotope. I would be shocked if a minimum phase ADC didn't sound better than a linear phase one. But all recording should be done at a high enough sample rate that an anti-alias filter is unnecessary. I suspect that there are more gains to be gotten from Redbook when people start doing things better on the record side, too.
> > I don't know Barry Diament < <
He's mastered many (link included). I have a few of his recordings, and they're well above average. Based on my rather limited exposure to his recording, he doesn't record hot, leaves lots of headroom, and the high frequencies seem more natural compared to most "screamer" CDs. His Led Zep III and Bob Marley Greatest Hits CDs, both of which I own, are easily the best I've heard of to these particular recordings.
TB1
It was partially a response to Peter Craven's claim than an apodizing filter eliminates all ringing from ADC. If that really was the case, then it almost wouldn't matter what kind of antialiasing filter you'd have to use in the resampler. So in that terms I agree with you. Although on paper the linear/steep filter is more extended, the minimum phase preset sounded more extended and didn't lose the timbre quite as much. As this was through the QB-9 which is minimum phase also. I can only conclude that minimum phase at the only DAC side is not quite the substitue for minimum phase at ADC + minimum phase at DAC, and therefore Peter Craven is wrong in that an apodizing filter eliminates ringing from the ADC. (although this is not an exact quote)
Or maybe he is right, maybe it does eliminate the ringing, who knows, but MP at ADC still sounds a whole lot better for some other reason, and that is all that counts.
I agree about high sample rates, all recording should be made at an high enough sample rate, but I was referring the times you need to downsample to Red Book format.
"therefore Peter Craven is wrong in that an apodizing filter eliminates ringing from the ADC."
Well, the proof that he is right is trivial.
But you have to quote the circumstances exaxtly as required: a minimum-phase apodising filter applied in the chain after the ADC removes
the pre-ringing of the ADC's linear-phase anti-aliasing filter.
bring bac k dynamic range
OK, I think I got it. Charles, I think that if you made it clearer in your white paper that the Listen filter is not an enhanced apodizing filter there would be less confusion on that point. Maybe there already is less confusion if you don't count me.
Anyway, thanks for the explanation Charles and Werner.
Tom
Hi Tom,
You're probably right. And if I were writing a college text I would have gone over it a dozen more time and hired a professional editor to tighten it up. But it's not bad for free! (I'm sure that many of our competitors are reading it very carefully.) Thanks for hanging in there until it made sense to you.
... for coming on a forum like this to discuss your work.
You've got my vote for Audio Man of the Year for making the MP filters an inexpensive upgrade rather than a new, more expensive line of players. That kind of character and commitment to your customers is pretty hard to find these days. I wish you great success.
Tom
"I would just like to be able to explain to people why players with the MP filter sound better. "
I don't think anyone can explain that.
bring bac k dynamic range
I got into a tangle with him over that....... I'll just give the link to the explanation............
....but maybe you're paraphrasing. I haven't read the white paper.
I'd like to ask Charlie H. (and others who believe the same) why you believe Red Book is a "compromise". What's wrong with a zero distortion and zero noise (recording) format, to start with ?Then, if you examine your audiophile CDs, you'll see that nearly *all* of them have dynamic range and signal to noise ratios of 95db - far more than a symphony orchestra produces. Many discs have even more (perceived) range if dither-noise shaping was applied.
Finally, the 44.1 sampling rate (as sourced from 88.2 or higher capture) covers the limits of human hearing. And *this* fact links up with what I'm hearing through Meridian's 808.2 CD player. It seems that the problem in the highs was playback's - not the format's - fault. The former harsh highs, full of "glare", are gone !!!
Edits: 09/19/09 09/19/09 09/19/09 09/19/09
"Finally, the 44.1 sampling rate (as sourced from 88.2 or higher sampling rates) covers the limits of human hearing."
So... Iffins 44.1 is the very embodiment of perfection, why would you need a higher initial sample rate???
Rick
As Werner says below, high-sample, high-bit rates were needed for production. And probably for sound quality if encoding wide dynamic-range music (like Mahler's Fifth).The format for *listening* is what I'm concerned with...and it seems that Red Book is high-resolution - as long as it was recorded higher. And this has been exactly the case since the early 1990's.....
Edits: 09/19/09
I don't know what "high-resolution" means. I too think that CD's can sound great but they are like records, marginal and few do. But most of mine are listenable and enjoyable and 1% of the hassle LP's were. But if they didn't have a marginal sample rate so many more systems and disks would sound good more of the time. What stinks is that the limit isn't even real, it's a business artifact.
And they don't cover the range of human hearing. Even as a lot of us grow long in tooth, we still have better transient and temporal resolution than CD's can provide. And kids hear well into the 20's.
I'm glad you like your new player and I bet it wrings out the last drop of performance of the medium. And that demonstrates the weakness of the medium, if it wasn't marginal, it would be easy to obtain that level of performance.
"Ear training is actually mind training, because the appreciation of sound is a learned experience and the more we experience, the more we learn. Although to our modern ears, Edison's acoustic phonograph gave a crude representation of the original, its first listeners felt that its reproduction was indistinguishable from real life. It is only with each advance in the state of the art of sound reproduction that people become aware of the shortcomings of the previous technology. For example, whenever I work at a very high sample rate, and then return to the "standard" (44.1 kHz) version, the lower rate sounds worse, although after a brief settling-in period it doesn't sound that bad after all."– Mastering audio, the art and the science, second edition (page 45), by Bob Katz
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Edits: 09/19/09
There isn't much evidence that "transient resolution" beyond Red Book's specs is needed. The limit of hearing 18-19kHz. Some young folks might hear to 20kHz, that's about it.RBCD, through the 808.2, sounds quite similar to 24 bit/88.2 signals (via DCS) to my ears.
And it's *never* easy to obtain good performance in playback !! LP is still improving some 60 years after it came out and over a hundred since the dawn of flat-disc technology.....
Edits: 09/19/09 09/19/09 09/19/09
1) You don't want to be at the mercy of the anti-aliasing filters built in the available ADC chips
2) There are overwhelming reasons to do production work at a much higher
resolution than the delivery format.
bring bac k dynamic range
It was merely an attempt at cynicism aimed at the OP saying: "Finally, the 44.1 sampling rate (as sourced from 88.2 or higher sampling rates) covers the limits of human hearing. And *this* fact...". Thought he might see the conflict between his statements even though he is wrong about the range of human hearing. At least for younger folks, sigh...
Obviously you should have adequate margins throughout the chain, including playback and CD's don't, at least not in sample rate. I think they are OK in dynamic range. Naturally you want margins when recording, but I see that as mostly a dynamic range issue to make sure that the capture doesn't clip and that noise from succeeding processes is inaudible. I suppose the same thing applies in time too, as you say to avoid nearby filtering at capture. Same reason you don't want to do nearby reconstruction filtering at playback.
I can understand people being contented with CD sound, I largely am. And the standard made sense in it's day. But that day is past and unfortunately the next generation was fumbled so we are still stuck with their HF compromises causing critical listeners to buy expensive gear that gussies up their defects. But that's a bandaid, not a solution.
Rick
I'd like to ask Charlie H. (and others who believe the same) why you "What's wrong with a zero distortion and zero noise (recording) format, to start with ?"
Nothing.
"Then, if you examine your audiophile CDs",
Funny. I don't have any audiophile CDs. I hope ...
"Finally, the 44.1 sampling rate (as sourced from 88.2 or higher sampling rates) covers the limits of human hearing. "
Yes, it probably does. For most human beings anyway.
And *this* fact links up with what I'm hearing through Meridian's 808.2 CD player. It seems that the problem in the highs was playback's - not the format's - fault.
"The former harsh highs, full of "glare", are gone !!!"
I don't have a Meridian Whatever.2, and yet I do not, and have not in the past decade, suffer(ed) harsh highs full of glare.
The reason CD is a compromise is simply in the fact that there are no prescribed, known-correct, rules as to how to band-limit the signal prior to sampling. And doing this in a transparent manner, but ascloseasthis to the edge of audibility, is not trivial. In fact, only a compromise is possible. Your Meridian picked one particular flavour of compromise.
"playback's - not the format's - fault. "
In digital audio one cannot separate recording, format, and playback. It is all one system.
bring bac k dynamic range
Are you saying that you don't hear an improvement with minimum-phase filters ? Please clarify...
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