Well written, interesting, provacitive. This coincides with my recent experience with forays into SACD & DVD-A. The overriding contribution to good sound (IMHO)is what I would call the production values( the care and equipment used in recordng & mastering). I have some dual sided DAD(24/96 PCM) and DVD-A (24/192 MLP) discs. I hear no advantage to the 192 sides. I consider the DADs proposed by Halverson & Classic Records to have been an ideal solution to improving the 16/44.1 standard, but of course it was never supported by the big players because it did not afford copy protection. I also find it very difficult to identify before purchasing "true" high def material actually encoded with 24/96 PCM, DSD, or even high speed analog tape with digital remastering. It seems to me these "high def" discs do offer improved sound, but only with the proviso cited in this article that the production values are also high def.
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I don't own any high res files, because none of the music I listen to is available in 24/96, but I am considering the purchase of several classical titles in 24/96 since I bought a DAC that can process the signal. The true answer to something like this is an A/B test between the same track at 16/44.1 and 24/96, and I plan to do this if I can. In the meantime, I can say that the 192kHz setting on my PS Audio DL III helps remove some high frequency noise and reduces listener fatigue. I am currently battling this HF noise in my DAC-direct-to-amp setup. But this has nothing to do with 24/96 files, all my stuff is Redbook...now I'm wondering if the 24/96 USB/SPDIF converter is inserting "audible intermodulation of the ultrasonics!"
I can only comment that I'd much rather listen to good 16/44 vs mediocre or poor hi res regardless of frequency, bit and sample rate... Although really good recordings on high res are wonderful...Recording quality is a higher factor. Than resolution once you get up to 16/44. I've posted a hi res quesion on the computer forum, like any feedback that I can get.
Even better than hirez formats are recordings produced correctly. If you want to hear great stuff listen to movie soundtracks. Somehow they almost always get it right. Even old stuff from the 50s can sound incredible, yet the music business can't seem to do the same with the same music. It seems the movie industry cares enough to get great sound while the music industry (whose job it should be to get music right) just does not care in general about sound quality. Maybe this is why there are so few new audiophiles today, because music is recorded and produced so poorly and sounds so bad that most never hear any need for something better than MP3s? I believe that the music industry is responsible for there own woes.
The benefit over 24/96 is usually subtle, but can be interesting in complex classical or female vocals. I have some 2L classical that is 24/192 and very nice indeed. The 192 titles are so sparse that its not really a requirement, only a nice to have in my book.
Some DACs however will sound a lot better due to the digital filtering automatically being pushed out beyond audibility with 192. I manually control this on my DAC, so I can do it even with 44.1, and I do. Sounds a lot better than a brick-wall filter at 20kHz.
Based on the article, I'm convinced and satisfied that there is no intrinsic benefit to hi-rez sampling rate. Although the quality of the recording, as always, plays a significant role in the quality of the audio files, its still independnet of the question of the resolution of the file. If/when I upgrade my DAC, I'll be looking for the best redbook dac. Such a shame that all the most 'up-to-date' fare will be contending with upping the sampling rate or other stuff that either has no audio benefit or just adds to or amplifies distortion.
I'm surpsied at some of the comments above suggesting that some of you guys still adhere to some benefit with hi-rez formats. I know the article is just one article, but I found it to be utterly convincing that there is no benefit.
I read that artical explaining why theroretically we can not hear anything above redbook. I think the key is to keep improving coding and decoding of this standard. It would seam that the team that developed the CD standard where no idiots knew exactly what the standard needed to be for the human ear. I just got a Metrum Octave last night and I must say it takes another (huge) step forward in extracting amazing sound from redbook recordings. Also, recordings have gotten vastly better in the the past 30 years. I now have sound quality that I believe truly rivals analog, and it only took 30 years to get it. he he
I agree with the comments above that embrace the view that the biggest problem is engineering and mastering. I've posted a number of "heads-up" comments in other threads when I came across a particulary good sounding red book CD. Same goes for vinyl. This is an industry problem and my concern is that the music industry will go after market share and demographics. Let's face it folks, the kids love those ear bud thingys and could care less about real music fidelity. Us "gray hairs" are a shrinking market niche. What's an audiophile to do??
yes, I seem to concur that 24/192 makes no sense. I never was a believer of the hi-rez scam that pervades the industry.
There is also a nice article written by Dan Lavry on Lavry Engineering's website on why sampling upto a max of 24/96 makes sense & anything beyond that is bogus. Dan Lavry wrote this article back in 2004! This article is called "Sampling Theory" when you go to this link:
Also agree with the above, there is only so much on a recording, mastering is where it's at.
My Dac is a custom built NOS AD1865K 16/44 with short signal paths, silver internal wiring and V-CAP OIMP output caps fed by a modded CEC transport, if there's anything missing it's the recording.
I have some (not all) quality redbooks that equal and beat some of my friend's LP counterparts from his Walker T.T. and we compare all the time.
This is a good thread so far.
Poorly recorded and mastered music doesn't sound good at any sample/bit rate. Is that really surprising?
The referenced article is nonsense. Essentially he's saying the math is perfect and therefore nothing else is needed for better sound quality. Sony/Philips said the same thing back in the mid-1980s. Out of bandwidth induced noise is a problem in both digital and analog. It is something engineers are well aware and there are a multitude of solutions with proven track records. If an audio amplifier manufacturer said his new amp filtered out all audio above 20kHz would you consider it a serious high end oriented design? If you really take the article seriously we should all be using 32kHz/12bit digital because the math works perfectly at that level too.
I'm sorry Onhwy61 the referenced article is not nonsense. We are dealing with a pure digital signal until the output of the D/A. So, there are plenty of DSP techniques available to make this work without oversampling the heck out of the digital signal. We need to oversample just enough to ease the specifications of the analog reconstruction skirt so that it's not brickwall. That's where 96KHz sampling comes in.
I also bet that most people's systems (including yours & mine) do not have 96dB of dynamic range after all the sweat that we have put in to isolate & reduce noise.
If you really take the article seriously we should all be using 32kHz/12bit digital because the math works perfectly at that level toono it does not. The Fs/2 freq would be 16KHz which would be less than 20KHz.
And, 12-b would be insufficient because one would add too much noise when going thru the mastering process & you'd effectively get 9-10 bits of music.
Just a little heads up on a well recorded redbook CD. I'm an old Four Seasons fan from the 60s and 70s. Just bought the CD 2005 original soundtrack of Jersey Boys and a CD redo of the original Best of the Four Seasons. The 2005 Jersey Boys CD is very nicely recorded. The original Four Seasons CD sucks. I guess the engineer who mastered the old Four seasons tunes thought everyone played their records with hand cranked turn tables and used a sowing needle for a stylus. UUuugghh.
And what's wrong with mathematically perfect response out to 16kHz? Most people, and certainly most middle age and older audiophiles' hearing doesn't go beyond 16kHz. FM doesn't go beyond 15kHz and at its best it's pretty damn good sounding. 12bit audio has a theoretical dynamic range of 72dB. That exceeds all but the most dedicated of audiophile rooms and systems. And if you increase the sampling rate above 32kHz won't you be introducing ultrasonic IM distortion?
44.1kHz/16bit is not needed. Done!
Some DACs however will sound a lot better due to the digital filtering automatically being pushed out beyond audibility with 192. I manually control this on my DAC, so I can do it even with 44.1, and I do. Sounds a lot better than a brick-wall filter at 20kHz.
This is exactly why 192 makes sense and why it (can) definitely sound better than Redbook...192 is way outside of the audible range of human hearing; what better "place" to apply the filtering...
There are some great articles on "pre-ringing" and jitter that address the reasons why HiRes (192 especially) has real benefits...
That said; a great Redbook recording does indeed sound...great!
I agree with Onhwy61 - article is a nonsense. First, motion that 16/44 is perfect if meets Nyquist criteria is first nonsense. Nyquist criteria applies only to continuous waves. Short high frequency bursts like cymbals will suffer the most of distortion. Second nonsense is that digital filter can suppress 96dB within 4kHz without any problem. There is no filter in the world, digital or analog, that can do that (no matter how many poles) with even group delays (or linear phase if you prefer). Uneven group delays will cause poor summing of harmonics (delayed differently) and change in sound. Reducing suppression won't help since low level signals above 24kHz will "fold" into audible band starting at 0Hz. Next nonsense is that ultrasonic frequency is harmful to the ear and modulate tweeter. Not only that 192kHz is WAY easier to completely filter out than 44kHz but also modulation can only happen on nonlinear element and for this to happen membrane has to move - not likely at 192kHz (even if your amp and CDP have such bandwidth). Then he claims that higher resolution does not increase dynamic range because of ambient noise floor forgetting that it is still improving resolution for louder signals. He claims that oversampling can increase resolution and sampling rate - true, but it is done with interpolated samples while 24/192 contains real samples. I agree that we might have hard time to hear better above certain resolution/rate, for instance 20/96 but claim that 24/192 is harmful is complete nonsense.
04-20-12: Onhwy61while it might be true that older ears do not have the 20-20K respone, the music is prepared for everyone. Like the article says there is a 100 yrs worth data that shows that 20-20K is the human hearing limit. So, when preparing digital music might as well keep the audio spectrum to its max limits. Younger people certainly can hear this range & so can many other older folks.
FM doesn't go beyond 15kHz and at its best it's pretty damn good sounding.FM has (air) spectrum bandwidth limitations that force it to curtail bandwidth. If they could help it, they would have also transmitted in the 20-20K range. Air spectrum is very expensive so this compromise seems reasonable.
FM doesn't go beyond 15kHz and at its best it's pretty damn good sounding.Like I wrote in my prev post & I'll write it again: if you start off w/ 12-b you'll end up with 9-10 bits after the mixing & mastering processes. If you start off w/ 16-b, you'll probably end up w/ 12-13 bits. The section "The dynamic range of 16 bits" explains quite well the DR of 16 bits & also how it might be possible to encode fainter signals using 16-b.
Since a lot of data already shows that sounds at absolute levels of +120dB, +130dB permanently damage ears, my understanding is that it might not be worth encoding sounds on a disk that cover the enitre 140dB dynamic range of human hearing. It appears that covering 120dB of dynamic range is sufficient. If one uses 12-b only & one attempts to encode very faint sounds my understanding is that 72dB could be a limiting factor trying to cover the entire 120 DR. 16-b & 96dB is adequate & the article shows a plot of a -105dB signal at 1KHz using clever dithering techniques.
Nyquist criteria applies only to continuous waves.nonsense! The Nyquist criteria applies to any signal that needs to be quantized. The Nyquist criteria only gives the minimum requirement; it does not say that one is forced to have only 2 samples per highest frequency.
There is no filter in the world, digital or analog, that can do that (no matter how many poles) with even group delays (or linear phase if you prefer).yeah, I know what you mean for analog filters & I agree w/ you in that respect but for digital FIR filters (linear phase) I'm not sure I totally agree with you. My understanding is that if you had a, say, 64-tap FIR you could have a very steep skirt digital filter that would have flat group delay & group delay distortion. I would have find some evidence of this before I contend this issue w/ you but for right now I'm skeptical that it cannot be done. I'll leave it that....
I see the case for upto 24/96 as it seems to alleviate most of the pressing issues such as noise creeping into the music signal during mixing/mastering, analog filters having too steep a skirt at 44.1KHz. I'm not sure that I buy the case for 24/192, etc.
If anyone is interested in looking at some signals look at the Powerpoint presentation at reference #17 in the article. SLides 20, 21, 24-28 show spectrum of instruments & spectra of music from commercial CDs. Look at the freq where the content dies off even for SACDs.
Short high frequency bursts like cymbals will suffer the most of distortion.Kijanki, this one is for you: here is a wonderful thread showing frequency spectrum of various brand of cymbals: http://www.drummerworld.com/forums/showthread.php?t=66957
The top quarter of the thread shows some really very good spectra of various cymbals. You can see that by 40KHz the spectrum has died down to 30-40dB SPL. The major part of a cymbal crash freq content is in the 20-20K range & the content falling off rapidly thereafter. I agree there is content beyong 20KHz but atleast 30dB by the time you hit 30KHz.
So, one could make the case for a 96KHz sampling rate wherein all the freq content upto 48KHz would be included. This sounds reasonable. At 48KHz the analog filter spec becomes reasonable too. Looks like it's a win-win situation....
Lots of technical info (on paper), but what really matters is how it sounds to each person. If you cannot hear that 24/192 WAV or FLAC sounds better than redbook CD, you may want to start selling off that expensive gear and get a boombox to listen to. My LINN Akurate DS playing 24/192 WAV or FLAC will sound better (everything else in the system being the same), than almost any CD player you put next to it.
i disagree with the article and the premiss behind it. nonsense sounds about right. i have nothing to offer in dispute of it except three years of listening to hi res (burned dvd's and streaming). the improved sound quality of hi res has been obvious to me in my set-up (given well recorded music from the start).
if you're not sold on hi-res.....fine. just don't tell me what i hear isn't for real and i'm "wrong".
04-20-12: Kijankinot true! Here is a link to paper written by Dan Lavry of Lavry Engineering who wrote this paper in 1997 that shows a 500-tap FIR filter that has a passband of 15KHz & a transition band of 1KHz & stopband starting at 16KHz. The attenution achieved in the 1KHz transition is a whopping 100dB!! See page 3 of 7:
yeah, it came at a price: 88.2 million operations per second using a dedicated DSP. Very high # of MOPS but do-able.
If one opens the transition band to 4KHz like the paper referenced by the OP then I'm sure that the # of taps will come down.
The paper also goes not to say that the group delay of the FIR filter is flat all the way out to 15KHz.
here is another digital filter paper (from the AES) that shows brickwall digital FIR filters:
it's possible to have these brickwall FIR filters with reasonable DSP capacity.
If you have not heard a properly setup DAC with hi-res files, you owe it to yourself to do so... before you spend any money on a CD player... you will be happy that you did. Not to mention the benefit seeing your entire music collection on an iPad and being able to make playlists, etc. Why would anyone want to search through a bunch of CDs, put it into a device that could only add noise and jitter and eventually will fail (drive, optics), and hear a few songs that you may like by one artist on that CD. Instead, if you like Chris Issak for example, have every song he every recorded in AT LEAST the same quality, and most times better quality sound than the std. CD, all at your finger tips. How can you beat that?
OK guys.... here is a link to this study, no wait.... here is a link to this paper, no, i mean, her a link to this test a dude did a while back that says you are wrong... etc. etc.....
SO WHAT!!!!!!!! How is all this arguing about studies and technical data and technical specs going to tell you how ANOTHER PERSON will interpret how a certain CD player playing a std. redbook CD sounds vs. a DAC playing a digital file !!!!!!!!!!!
"nonsense! The Nyquist criteria applies to any signal that needs to be quantized. The Nyquist criteria only gives the minimum requirement; it does not say that one is forced to have only 2 samples per highest frequency."
Yes, you can have more samples (for instance 192kHz) but he claims that 44.1kHz (two samples) is all you need.
Again, Nyquist applies to continuous waves ONLY.
"The theorem assumes an idealization of any real-world situation, as it only applies to signals that are sampled for infinite time; any time-limited x(t) cannot be perfectly bandlimited."
Perfect reconstruction of continuous signals close to Nyquist frequency (for instance 15-20kHz) is possible but when signals become very short, reconstruction is much less than perfect.
As for filters - look at typical response of 2and 8 pole 20kHz Bessel filter in dB:
20kHz -3 -3
22kHz -3.63 -3.67
40kHz -9.82 -13.68
80kHz -20.32 -51.81
As you can see there is very little attenuation difference at 44.1kHz/2=22kHz with 4x higher number of poles. You would perhaps need hundreds of poles and still not get -96dB. Dramatic difference shows at higher frequencies beyond the "knee" of the filter (160dB vs 40dB per decade). Whole purpose of converting analog to digital at higher rate is to represent bandwidth of 20kHz more accurately and not to extend bandwidth. Downsampling 24/192 master tapes to 16/44 removes some information, (audible or not) but to claim that 24/192 is inferior to 16/44 is complete nonsense.
As for dynamic range, again the point is resolution of the signals above noise floor. According to this article if I listen at 85dB peak and have 35dB ambient noise at home I should not be able to tell the difference between 16 and 8 bit recording (corresponding to about 50dB range). That's nonsense as well.
What about 192kHz being harmful? It doesn't get more silly than that.
The article seems right, as far as the author goes. He admits to the problem of brick wall filters on Redbook CD, but he forgets to mention timing errors. This is why a cirrectly clocked computer regenerated waveform seems to improve on the CD, thru the same DAC. I would propose a much reduced timing error as part of the improvement of the higher res sampling frequenies. There is also a lot of talk on WAV beng better than FLAC (even though "bits are bits").
Stretching, I would hypothesize that the regeneration of the waveform by the additional complication of the FLAC decompression "bothers" our sensibilities in some way. IF so, then I would also propose that the decompression of MLP on a DVD-A might be similar.
My hypothetical ranking of sound which seems to agree with what I hear is:
CD<=FLAC<=WAV (for 16/44.1)<=24/96MLP<=24/96PCM<=24/192MLP (for DVD/A <= DSD, and just for fun <= LP. At some points the minor improvements may not be worth the additional storage requirements.
Remember, just speculating!
What a great thread! Civil discussion. Ruffled feathers, for sure. But, civil and respectful. Keep going guys! I am still on the fence about hi-rez and the OP really brought up a great article and it sparked a great discussion. It will ultimately come down to objectivists vs subjectivists, but who cares? Those who dismiss A/B/X testing believe that there is more to listening than can be quantified or measured, and those who believe in A/B/X think the rest of us are fools if we don't follow the science. Thanks to all for an entertaining discussion!
All this talk about tests and links to this study and all this technical stuff....
REALITY: All that means nothing when you have people that like and dislike things based on their own opinion of what "sounds good". Many feel tubes sound better than SS.... HOW CAN YOU TEST THAT? YOU CANNOT. The listener has their opinion of what sounds good. So posting links to these different things is POINTLESS.
On the question of continuous vs. non-continuous waveforms, I think that part of the reason for the disagreement is that the word "continuous" is misleading in this context. No waveform is truly "continuous." Regardless of the nature of the waveform, the Sampling Theorem will only be perfectly accurate (i.e., to 100.00000000...%) when an infinitely long sample record is available, covering the period from the beginning of the universe to the end of time. :-)
Any real-world waveform, whether sinusoidal or not, and "continuous" or not, will not meet that criterion. As a result there will always be some non-zero loss of information, at and near the times when the waveform begins, when it ends, and when it changes character. In theory the spectral content of those transitions extends out to infinity Hertz, although as a practical matter much of the high frequency spectral content of those transitions will be at amplitudes that are utterly negligible.
The information that is lost in those transitions will correspond to the spectral components that lie above the cutoff point of the anti-aliasing filter. The lower the cutoff point of the anti-aliasing filter, and the more abrupt the transitions are in the waveform that is being sampled, the greater the amount of information that will be lost.
Will any of that particular form of information loss be audibly significant when a music waveform is sampled at 44.1 kHz? It's hard to say, and I doubt that empirical assessment (by listening) can yield a meaningful answer considering how many other variables and unknowns are involved in the recording and playback processes. My guess is that it probably has some significance, especially on high frequency transients such as cymbal crashes, but only to a relatively small degree.
Is oversampling plus noise shaping an essentially perfect means of overcoming the problems inherent in sampling just slightly above the Nyquist rate, as the article seems to suggest? It's probably fair to say that it can work pretty well, but IMO it would be hard to argue that it is "essentially perfect." Can the ultrasonic frequency content that is retained by hi rez formats have adverse consequences, as claimed in the article, as a result of intermodulation effects within the system's electronics, or things like crosstalk effects for that matter? It certainly seems conceivable, to a greater or lesser extent depending on the particular components that are in the system. Will sampling at a higher rate result in sampling that is less accurate, assuming equal cost and comparable design quality? That would seem to be a reasonable expectation. But complex and sophisticated digital signal processing does not come for free either.
What does it all add up to? I would have to say that the paper referenced by the OP, and also the Lavry paper, make better cases against hi rez than I would have anticipated, but they are certainly not conclusive as I see it. And given the many tradeoffs and dependencies that are involved, my suspicion is that there will ultimately be no one answer that is inarguably correct.
Al, What sounds inconceivable to me is that 24/192 recording supposed to gain sound quality by downsampling it to 16/44 to be upsampled again, perhaps to the same 24/192. Am I reading it right? Is downsampling + upsampling somehow improving sound by replacing real samples with artificial interpolated samples and recreating same harmful 192kHz?
The only reference to downsampling + upsampling that I recall seeing was in the paragraph headed "clipping" in the lower third of the page, and in footnote 21. He was saying that by taking 192 kHz source material, downsampling it, and then upsampling back to 192 kHz, a sonic comparison could be made between the two 192 kHz signals that would be indicative of the adequacy of the lower sample rate.
Not sure if that is what you are referring to. But in any event the methodology he is describing doesn't make sense to me, because the comparison would not reflect the effects of the sharper anti-aliasing and reconstruction filters that would be required for recording and playback at the lower sample rate.
Al, On one hand he has whole chapter titled "192kHz considered harmful" describing harm that 192kHz can do to amplifiers and speakers to later say this about oversampling:
"This means we can use low rate 44.1kHz or 48kHz audio with all the fidelity benefits of 192kHz or higher sampling (smooth frequency response, low aliasing) and none of the drawbacks (ultrasonics that cause intermodulation distortion, wasted space)."
According to above when DAC is exposed to 192kHz sample rate from CD it is harmful to all analog circuitry afterwards (including power amp and speaker) but when DAC upsamples redbook CD (24/192 downmixed to 16/44), the same 24/192 becomes benign. In either case DAC outputs samples at 192kHz rate but in one case it is less harmful to analog circuitry?
Ok, I see what you are referring to. Note this statement:
Oversampling is simple and clever. You may recall from my A Digital Media Primer for Geeks that high sampling rates provide a great deal more space between the highest frequency audio we care about (20kHz) and the Nyquist frequency (half the sampling rate). This allows for simpler, smoother, more reliable analog anti-aliasing filters, and thus higher fidelity. This extra space between is [sic] 20kHz and the Nyquist frequency is essentially just spectral padding for the analog filter.So what distinguishes the two situations you are referring to is that the 192 kHz hi rez format will presumably include a significant amount of ultrasonic audio information, which is what he is saying might have harmful consequences as a result of intermodulation effects in downstream components, while the oversampled redbook data will not include that information, and therefore those effects will not occur.
04-20-12: KijankiKijanki, all that the author is saying is that when sampling at higher freq like 96KHz or 192KHz, you get intermodulation products that fold down into the 20-20K audio band due to typical preamp, power amp bandwidth limitations of not being able to reproduce higher freq products distortion-free i.e. due to the non-linearities of the electronics. And, systems having smaller bandwidths have the situation worse in that the probability that they'll amplify the high freq signals is much higher. So, the point is that if you do not sample at 96K or 192K you won't have these higher freq intermod products, they won't fold down into 20-20K & your preamp/power amp will not amplify them due to its non-linearity.
it's clear to see that if a 96K or 192K sampled signal is downsampled to 48KHz then the anti-aliasing filter will cut off all these high freq intermod products. So, according to the author, since this signal is free of any ultrasonic content, it's safer to playback with the idea that distortion products due to ultrasonics are not being played back.
I do not think that it's unreasonable to say that ultrasonics created due to higher freq sampling can create in-band intermod products that can be amplified by the non-linearities of the playback electronics & that they are harmful to the playback listening pleasure.
I don't think that the author should have labeled the paragraph as "192KHz considered harmful". People like Kijanki have read this literally thinking that the very act of sampling at 192KHz is harmful. No, I don't think that the very act is harmful; it's those ultrasonics folded down & amplified that are harmful.....
Suppose we want to compare the fidelity of 48kHz sampling to a 192kHz source sample. A typical way is to downsample from 192kHz to 48kHz, upsample it back to 192kHz, and then compare it to the original 192kHz sample in an ABX test .Al, Kijanki: I *think* that I might know what the author is intending to say here: To do an A/B comparison, the author would like to level the playing field. Thus, he does not want to use the original 192KHz signal as-is. What he wants to do is downsample on-the-fly the 192KHz signal to 48KHz & create signal A. Then, upsample this 48KHz signal on-the-fly back upto 192KHz & create signal B. Now, the playing field is level because the same machine downsampled & upsampled the signal & the same filters have screwed up the A & B signals. The signal X is the original 192KHz signal. If you were to use the original 192KHz which was created on some different machine against the 48KHz created on your CDP, you would have the effect of 2 different digital filters & you could not do a true A/B comparison. Does this make sense guys?
if you're not sold on hi-res.....fine. just don't tell me what i hear isn't for real and i'm "wrong".if you are sold on hi-rez, fine. I don't know what you are listening to - true hi-rez (which would mean the analog masters sampled at 96KHz or 192KHz & made available for purchase) OR bogus hi-rez (whcih would mean taking the 16-b CD data, resampling it at 96K or 192K & providing it for sale to the unsuspecting public).
There have been sooooooo many scams re. hi-rez (recently read something about HDTracks were the offenders. Here is the link:
that it's really very difficult to tell what the manuf has provided for sale.
You are probably listening to some digital filter thinking it's hi-rez & you are in 7th Heaven.
Maybe we should let you be - ignorance would be a bliss for you.....
As for filters - look at typical response of 2and 8 pole 20kHz Bessel filter in dB:Kijanki, you provided us w/ the freq resp of ANALOG Bessel filters. I agree with you & I did write this in my prev post - analog filters cannot creat a sharp cutoff like what the author has shown - large attenuation between 20K-24K.
But, how about digital FIR filters? Can they create such a sharp roll-off?
Yeah, sure they can! Did you bother to read any of the links I referenced in my post? The paper from Dan Lavry shows 1 example & then there is that AES paper by Julian Dunn that also shows 4 filters that have 100dB atten & only a modest # of taps. All FIRs have flat group delay in-band.
I found this nice article called "A Beginners Guide to High Resolution Downloads of Music". here is the link:
In this article is a para called "How hi-res should you go?" towards the bottom (scroll almost to the end). That para cut & paste:
'Unless you have extremely youthful hearing ability plus the highest-end speakers and audio gear, many of us feel that the improvement of 192K over 96K is inaudible. The word length expansion from 16 bits to 24 bits makes a much greater enhancement in the sound. 24/96 or 24/88.2 is fine for nearly everything. Also, remember that 192K and 176.4K files take up much more memory on hard drives, for little audible improvement.'
Looks like many people think alike: there's a case for 88.2K or 96K sampling but not beyond......
I think that what he is referring to in note 21 and in the "Clipping" paragraph is a comparison between a 192 kHz hi rez signal, and that same signal downsampled to 44.1 or 48 kHz and then upsampled back to 192 kHz. Both 192 kHz signals would be played back through the same DAC and the same downstream components. If they were to sound different in any way it would presumably mean that the lower sample rate, and/or the downsampling and upsampling processes, degraded the signal. Which signal sounds subjectively better would be irrelevant.
As I indicated earlier, though, it seems to me that the flaw in that methodology is that it does not take into account the sonic effects of the anti-alias and reconstruction filters that would be used if the recording and playback processes were done at the lower sample rate.
put it into a device that could only add noise and jitter and eventually will fail (drive, optics), ....hey Audiofreak32, don't talk about hardware failures! With your being sold on computer playback you don't have a leg to stand on when it comes to hardware failures! How often does computer hardware fail compared to CD drive & its optics? Even the cheapo $40 DVD players from Walmart outlast almost all HDDs & other computer hardware.....
Yeah, the convenience of HDD playback is immense, I have to agree.
If you have not heard a properly setup DAC with hi-res files, you owe it to yourself to do so...I have - dedicated computer for music playback, going into a dCS upsampler, going into a dCS DAC. Both dCS upsampler & DAC were clocked by a dCS Verona Master CLock. All interconnects were some very expensive WireWorld stuff. The total $ outlay on this whole setup made my knees weak - I could never afford anything like this for a long time! The sonics were easily beaten by my 1-box CDP.....There was no body or soul to playback MUSIC but the SOUND was stellar.
Al, I wonder if 24/192 contains any ultrasonic frequency at all. Why would they leave it preparing hi-rez files? Where this ultrasonic frequency comes from? Again, notion that 192kHz sampling is harmful is a little farfetched. Do we have any studio sound engineers on our forum that could explain it to us?
Bombaywalla, Thanks for the info on filters. I'm dealing mostly with 4-tap lowpass FIR filters at work but 500-tap filter is really something. One graph shows interesting step response typical to most of CDPs with ringing appearing before and after the pulse. That might affect the sound since our ears are very sensitive to it. Stereophile posted similar test results comparing apodizing and non-apodizing filters. In comparison there is no antialias filters used in SACD creation making better, more natural step response (transients).
I do not have golden ears and like the sound of my system very much but just believe that processing back and forth 24/192=>16/44=>24/192 is not likely to improve anything. Higher sample rates are not to extend bandwidth but rather improve filter response reducing pre-echo effect. Apodizing (windowing) filters, available in few CDPs like Meridian, allow to eliminate pre-cho completely but AFAIK are not suitable for 44.1kHz because there is not enough space between 20kHz passband and first alias to fit filter's windowing function. DSP processing is not my field of expertise but even if everything looks peachy in frequency domain there is a lot to be improved, possibly by higher sampling rate, in time domain (transient response).
Audiofreak32, technical articles are to understand better what is happening but you're right, that at the end what counts is listening experience. At the level of 20/96 or 24/192 placebo (or negative placebo) effect might be a dominating factor. Just the fact that I feel good about my gear can make it sound better to me than to others.
Oldears, Choice or audibility of different formats might depend on setup. In my setup, for instance data is wirelessly delivered ALAC compressed to Airport Express and contains no timing. It is also bit perfect. Lack of timing is important because it eliminates any influence of computer processing or playback program, computer noise, etc. At this point timing is recreated in AE and data is streamed to Benchmark DAC1 with low 258ps jitter further suppressed by Benchmark processing. I could also save data in other formats but it would eat up some processing power of my computer that I use for other chores (like typing this).
A LINN Akurate DS is around $7,500 new. Only other things you need are a NAS and an iPad. So, less than $10k retail, easily. Sure that setup you described was expensive, but you do not need a "dedicated computer" at all. I am using a $350 NAS with a 3TB HDD loaded with 24/96k and 24/192k WAV and FLAC files. The only IC's I need are a pair of RCAs into my amps. So, I am talking about a DAC (no moving parts) and yes, a HDD in a dedicated NAS, but compared to the alternative? Really?
Audiofreak32, I enjoy my system all the time. ALAC is lossless while wireless transmission is bit perfect. Benchmark is as clean as it gets on the verge of being sterile but it fits perfectly with my warm sounding Hyperion HPS-938 speakers. In addition my Benchmark is modified with better sounding op-amps. Sure I could be doing much better but for much more $$$$.
Linn looks very impressive but it is $7500 while Benchmark + AE were $1100 total. I'm perfectly happy with 16/44 limitation of AE, having over 1500 redbook CDs on HD. Computer costs me nothing since I already have one. My setup also requires only one pair of ICs.
04-20-12: KijankiHi Kijanki,
What he is referring to is the ultrasonic output of the musical instruments themselves. Yes it would be at very low levels, and with a lot of instruments it would probably not be present to a significant extent at all. But his point, debatable though it may be, is that leaving it in can't do any good, and MIGHT do some harm, depending on the non-linearities that may be present in the playback system.
It would be left in the hi rez recording to avoid introducing a sharp cutoff filter into the signal path, which as you realize is one of the fundamental benefits of high rez.
Along the lines of my earlier comments, I'm skeptical and/or uncertain about a lot of his points, and how they would trade off in terms of significance against the presumable benefits of high rez. But I don't consider his arguments to be outlandish or unreasonable.
Outlier - thanks for posting the liink. Very interesting and helpful article - no matter what side you come down on in the hi res debate. Thanks also to Bombaywalla for the "beginners guide" link. Am I missing something or isn't the convenience of computer-based audio going to apply whether or not music files are hi res?