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In Reply to: This Engineering Publication Gets It Right..... posted by Todd Krieger on February 12, 2007 at 04:11:16:
I suppose with the image of engineering that the smug sales/marketing folks end tends to give, it will be a surprise for some to realize that in some cases, engineers actually know what they’re dealing with (designing the stuff) and are some are not even deaf.
EDN is a great mag btw, they used to have a cool series by a guy named Robert Pease (sp?) who was a real circuit wiz.
Best,
Follow Ups:
Hi.Last month my audition between the CD 44.1KHz 16-bit & DVD-audio 196KHz 24-bit formats of the SAME identical sound tracks of two total different recordings (from two different labels), has proven to me the sonic difference is nite & day.
Needless to say, 196KHz 24-bit blows alway its historic counterpart, in term of 3-D spatial presentation alone, not to mention other goodis, e.g. tonal & resolution improvment.
If any armchair experts claim this was engineering gimmicks, I will hereby dismiss such allegations as poor aural perception, hearsays,
or undisclosed motives or agenda.FYI, I got music reference CDs from the same recording/produciton lab
employing the same microphones & recording gears, on 16-bit & 24-bit mastering. I can distinctly hear the 24-bit mastered CDs sound so much more musical.Hearing is believing, assuming I am a scientific illerate. But I am not that ignorant.
I only wonder whoever made such allegation has actually tried out what I have done.
c-J
They hear the improvement of the better formats, and so do you.
I wish, that all people writing about this subject, would go to a recording studio and listen, before they write.A reality check is the only valid test of any theory in science.
A small nitpick, it is 192kHz, not 196 kHz.
> A reality check is the only valid test of any theory in science. <You're saying we should LISTEN????? Arrrrrrggghhh!!!! This is AUDIO, man! MEASURE it, don't listen to it!!! Sheesh! You must be new around PropHead. :)
Now, on to more serious concerns. I've done what you've suggested and yes, there are differences. On the other hand - and I won't mention any names - I've also reviewed one of the dual layer SACD/CD discs. The SACD layer sounded so much better but the CD layer was bad, bad, BAD! Worse than the original CD pressing. It almost sounded like they purposefully made the CD layer sound bad in order to tout SACD. In this case, the test called for in this thread would appear to be valid and a simple comparison not so.
I've done a little test once:
Somebody on another forum recorded a hihat and there were 5 files all in 16/44.1, all recorded using the same equipment (ie same mic, same mic pre, same convertor).
The difference was that one or more were recorded straight at 16/44.1 whilst one or more were recorded 24/96 and then dithered down to 16bit and resampled to 44.1.
The files were not marked in any way, the only way to tell any difference was by listening.
And despite the resulting files all being 16/44.1 the difference was subtle yet clearly audible.
If its recorded in 24/96 it does sound more realistic even if it ends up on a redbook cd.I guess there are some misunderstandings about Nyquists theorem.
It states that it is POSSIBLE to recreate a sampled waveform accurately if the sampling frequency is at least twice the highest signal frequency.
This does NOT mean that accuracy is guaranteed!
Accuracy is only guaranteed if the peaks of the original waveform coincide precisely with the instant the sample is taken. This instant occurs roughly halfway thru the sampling process and is very much shorter then a 1/44100 of a second.
Basically the sampling circuit consists of a switch and a capacitor and during the process it goes through three discrete steps: Firstly the switch opens to drain the previous charge, secondly the actual sample is taken and thirdly the switch closes and the voltage value is read out. All these steps each take a finite amount of time which means that the sample is not averaged over 1/44100 sec as this is technically impossible.
Four samples per waveform period are the minimum to reproduce the waveform with any degree of accuracy, eight would be better otherwise the sum truncation error can become very large indeed! Unfortunately this problem does not show up on standard distortion measurements as this only shows up anything that has been added (ie output - input = distortion). Since sampling will always leave things out rather then adding the resulting distortion would be negative (provided everything works perfectly which it never does).
That said Mister Lavry produces some of the best medium sampling rate convertors around, still I side with George Massenburg on this one.
Howdy
You said "I guess there are some misunderstandings about Nyquists theorem. It states that it is POSSIBLE to recreate a sampled waveform accurately if the sampling frequency is at least twice the highest signal frequency. This does NOT mean that accuracy is guaranteed!"That's actually wrong. It more accurately says that in the ideal world you will exactly reconstruct (no matter where the samples land) any bandlimited waveform. There is still a lot of wiggle room there, but not as you state.
Some of the problems are building perfect anti-aliasing and reconstruction filters. Other problems are that truly bandlimited signals have properties that may be counter intuitive. But with proper filters and a bandlimited signal every freq up to the Nyquist rate are reconstructable even if there are almost only two samples per cycle.
This subject has been beaten to death here and else where, but reading a real book on sampling theory should help to make it clear.
Ted,
you are absolutely right about Nyquists theorem. Of course when it says 'bandlimited waveforms' this actually means that it is not possible to reproduce a square wave of any frequency as square waves contain all harmonics at equal level. Personally I like my kit to be able to reproduce any waveform successfully, at least within the range of human hearing.
And I don't know about you or anybody else but I can quite easily hear the difference between sine, sawtooth and square wave up to at least 16kHz (thats were my generator cuts out).
It is also true that I have not read the "real book on sampling theory"; my sources here are a lecturer at a technical college (his subject: Digital Audio) and my uncle (Phd in higher mathematics and > 10 years of R&D for the European Space Agency specialising in all things digital ie digital data transmission, fractals etc).Either way I can't really see how it would be possible to recreate a sine accurately if during the recording process the actual peak values are missed. You can't recreate what has never been encoded. By the way last time I looked the 'reconstruction filter' was nothing more then a slew generator which leads to a phase delay which tends towards 90deg as the 'two samples per cycle' situation is approached. Feel free to elucidate...
and I wondered about a similar point myself. If you sample a sine wave at EXACTLY twice its frequency, it is possible to get all zero values. And obviously, this tells you nothing about the signal. Here are a couple of answers:First, the Nyquist theorem has, I believe a greater than sign, not greater than or equal to. And if you are even slightly greater than twice the frequency, you will eventually hit a peak.
Secondly, Pohlmann addresses this exact question in his book Principles of Digital Audio. His answer, to paraphrase from memory, is that, in practice, you are sampling complex signals that are not sine waves and this issue is never important.
> Either way I can't really see how it would be possible to recreate a
> sine accurately if during the recording process the actual peak values
> are missed.Consider sampling a single sine wave 4 times per period starting at 0 degrees:
0 +1 0 -1 0 ...
or starting at 45 degrees:
0.707 0.707 -0.707 -0.707 0.707 ...
The magnitude of 1.0 is contained in the information in second series even though the peak value never occurs. This is because the signal is not reconstructed by joining up the dots but by integration over orthogonal functions.
If you consider the coefficients of a DFT for the first series the one involving a sine wave at a quarter the sampling frequency will sum to 1.0 (ignoring scalng) with all the other sine and cosine coefficients summing to zero.
For the second series you will get non-zero coefficients for both the sine and the cosine coefficients at a quarter the sampling frequency with the other coefficients summing to zero. This will be a sine wave of magnitude 1.0 shifted by 45 degrees.
Of course a series like 0.707 0.707 0.707 will never arise unless the frequency is exactly half the sampling rate which is impossible.Also how does the D/A convertor know that the original was a sine wave when the only information its got is a numerical voltage value?
Its easy for you because you got prior knowledge, there are no such luxuries for the convertor. Could have been a sawtooth, ramp, triangle or square wave.
Or worse still a complex, non repetitive audio signal.
HowdyHere's an 18KHz Sine wave (in a 16bit 44.1K sampled signal.) The little squares are the samples and the green waveform is the (unambiguously) reconstructed sine wave. Obviously there are fewer than 3 samples per cycle (44.1/18 = 2.45 to be precise).
If you wish you may specify any band limited wave you want and I'll show you the unambiguous reconstruction.
> Of course a series like 0.707 0.707 0.707 will never arise unless the
> frequency is exactly half the sampling rate which is impossible.It was an easy to understand example to show why your thinking was wrong. Analogue signals are not reconstructed from digital ones by joining the dots. (The example was quarter the sampling frequency because half the sampling frequency aliases - DC and half the sampling frequency give the same sequence.)
> Also how does the D/A convertor know that the original was a sine
> wave when the only information its got is a numerical voltage value?I take it from this question that you do not know what a DFT is or how a D/A convertor works. I will pass on giving the lecture since it easy enough for you to look up on the web.
> Its easy for you because you got prior knowledge,
A DFT is linear and so will only ever give one answer. Prior knowledge of the answer does not change the answer.
> there are no such luxuries for the convertor. Could have been a
> sawtooth, ramp, triangle or square wave.These would all give a different series. They also all contain an infinite number of frequency components as mentioned by Ted and so cannot be represented exactly by a finite number of samples.
> Or worse still a complex, non repetitive audio signal.
Again we return to your not understanding how to transform samples between time and frequency space and what information is contained in the samples and what is not.
HowdyYou said "... Could have been a sawtooth, ramp, triangle or square wave." But this isn't true. It had to be band limited before conversion so those don't apply.
That's part of what I was trying to get at by stressing the band limiting in my first post.
Also a proper reconstruction filter is the same as a proper antialiasing filter: ideally it passes all freqs less than the Nyquist rate unchanged and eliminates all higher freqs. There are better and worse implementations (approximations) to this in practice and you described one of the worst.
"Also a proper reconstruction filter is the same as a proper antialiasing filter: ideally it passes all freqs less than the Nyquist rate unchanged and eliminates all higher freqs."Actually, if you go back to Shannon's proof for the theorem you'll find that the reconstructor is unambiguously defined, and that the AA filter is not defined at all.
--
The reconstructor, for the theorem to hold, has to be our infinitely-wiggly f(r)iend Sinc(x). If applied in all its glory the wiggles cancel out completely, so it has no drawbacks in the time domain (despite contrary claims from, oh, the Wadias and the Todds).
Small detail: most silicon-based reconstructors aren't quite like Sinc, but are half-band low-pass FIRs, with only 3dB attenuation at
fs/2.--
The AA is not defined. It is up to humanity to come up with an AA filter that guarantees zero aliasing all the while being transparent to the human ear. If you use Sinc(x) or a wiggly FIR here you introduce wiggles into the recording. These are there to stay.
And again, most silicon-based AAs are half-band FIRs, even the ones in 'pro' gear.--
Much more interesting for AA duties are linear phase low-pass filters with a strictly monotonic frequency curve (no ripple) and a wide transition band. If you want a (more or less) flat extension to 20kHz then the latter can only be supported by using a higher sampling rate. In the case of 96kHz you would keep everything flat to 20kHz (or rather, slowly dropping to -0.1dB at 20kHz or so), and then slowly roll off to -140dB at 48kHz. Such filters still introduce time-domain wiggles, but they are much shorter/faster than conventional brick wall filters.
See Dunn, Craven, ...
--CD's Catch-22:
1) We can only hear out to 20kHz.
2) Nyquist/Shannon are right and fs > 40kHz perfectly captures a 20 kHz band-limited signal.
3) So 44.1kHz is fine.
3) There is no way for transparently limiting a signal to 20kHz while using a 44.1kHz sampling rate AND keeping the in-band frequency response ruler flat.
5) So either 44.1kHz is not sufficient OR the system's frequency response should be allowed to roll-off through the upper octaves.
HowdyThanks for the correction RE the anti-aliasing. I tend to think of the pure audio world and the simplification of both filters being the same works well, at least in the abstract. Obviously when you get to best practices things are different...
-Ted
P.S. Why didn't you jump in earlier when people were making a mockery of Nyquist rather than picking on me :)
Hi Ted,not picking on you, not at all.
And why not earlier? Well, I only visit PHP twice a year ;-)
Back to business. All too often the reconstructor is specified as a filter (any filter?) with flat in-band response and infinite rejection past fs/2, and the specification for the ideal (ideal according to whose requirements?) AA filter. But specifying the reconstructor like that is wrong, or at least misleading. And copying the spec for the AA filter is wrong, as it is good for Nyquist, but not necessarilly so for the application at hand.
bring bac k dynamic range
HowdyWell that was my point :)
The op claimed that you needed more samples and that you could only represent signals whose peaks got sampled exactly. My point was that there are counter intuitive (to the uninitiated) restrictions caused by the requirement for band limited input (e.g. between 1/4 and 1/2 Nyquist only sine waves can be represented, no triangle, square, ...) and even more so the fact that there are no perfect implementations of the required filters. I was pretty careful to talk about ideal vs. actual implementation being a problem.
Oh well, back to "The Nutcracker" in multichannel on SACD which doesn't have these particular problems :)
Ted and Werner - Excellent discussion! Thanks so much...
Leaving something out is the same as adding (the same thing) 180 degrees out of phase. Distortion measurement shows either with just the same effectiveness.
Hi.That is exactly what I said in my post above.
I've acquired quite a few reference CDs with stereo sound tracks recorded by same mics & digital recorders, most of them mastered in 16/44.1 & a couple in 24/96 but down-converted to 16/44.1, produced by the SAME music production lab. Being reference music CDs, there a complete list of equipment used in the digital recording, layout map of the music instruments & performers, & description of the recording process.
I can hear 24/96 is more than subtlely superior to 16/44.1 with these reference music CDs.
As a music lover, I won't give a rat ass to get to the bottom of WHY such higher-bit+extended-sampling-frequency format does improve the sound. I go for better sound that I can afford. DVD-audio is the answer for digital sound todate. I really wish Blu-ray audio will take over one day given the maturity of the marketplace in this new technology.
Like I always find Wendy's burger tastes better than McDonald's.
As a consumer, do I need to find out why it is so????Wendy is not McDonald's despite theg both supply burgers. Do I have to summon the principle chefs of these two fastfood chains to cook a live demonstration on the SAME recepes in front of the world, to make sure which of the two chefs were the better cook?
I read some posts down below being so off-the-track that make me laugh.
The EDN paper does not impress me at all. It argued that there SHOULD not be aural difference using the century old cliche data, like our very limited 12.5KHz hi-cut auditory response, "a few mS" slow phase/time delay response, blah blah.
If our aural perception was merely defined by 12.5KHz hi-cut reponse & "a few mS" phase/time delay resulation, our technical revolution on audio electronics would be a redundant waste.
Where are you guys common sense & logics?
What we don't understand how our brain interpretes music today doesn't not stop us from believing what we perceive now. Time will tell. Who needs those so called engineer's papers cast their vote on our hearing on hollow arguments.
Even the Times magazine last month published a report of the latest survey on re-mapping our brain. Time changes our understanding.
c-J
Hi.ignoring music is for listening since day one of human civilization.
When mom says refrain from fire, it will burn your fingers. The toddler talks back: really? Show me how?
xc
xx
The threshold for disproving something is higher than the threshold for saying it, which is a recipe for the accumulation of bullshit - Softky
Hey CJ:Wassup?
I hate to do it man. But I gotta concur with Andy19191 and theaudiohobby on this one.
Try Natalie Merchant DVD-A versus CD. Listen to the bass line on the "Carnival" track and tell me those two products are from an identical mix.
Although it would be novel (hell - AWESOME) if we could know whether or not the same masterING was used for each resolution, I think it's really hard to know for sure.
Now if we're talking a simple digitization of an ANALOG master tape, well that's a no brainer! I want 24/96.
24/192 is just too problematic right now (because I am doing the PC audio thing).
Provenance of the discs themselves?
Different lasers and different electronics in the playback device?
Use the same DVD-audio player to play both the DVD-audio & CD discs, most likely single tray, introducing time lapses between comparisons?
Psychology? What's that?The true subjectivist audiophile would not allow any of these trivialities to intrude upon reality.
Here are the real facts.
"They" hear no stunning improvements.
1) Their system lacks enough resolution to hear any differences.
Most likely culprits are glare and grain. Usually both.
2) Their aural perceptive abilities are weak to nonexistent, aka deafness.
Possibilities include the ingestion of too many hamburgers or putty ear syndrome. Plus a host of others, too many to mention.
All of which leads to:
3) Jealousy. Envy. Faulty Logics. Probably from taking classes like static and dynamic.Heed not the perpetual doubters.
May the farce be with you.cheers,
AJ
The threshold for disproving something is higher than the threshold for saying it, which is a recipe for the accumulation of bullshit - Softky
POLLYinFLA is at it again! His pompousness is repeating anything he hears in typical objectivist parrot fashion!At least this time the bumbling bird brain has spoken some truth, which is an amazing thing in and of itself, although he doesn't even realize he's done so! I guess if you repeat what you hear often enough it was bound to happen eventually. So what is this truth his pompousness, POLLYinFLA pronounced?
Quite simply it's this...
Here are the real facts.
"They" meaning Objectivists hear no stunning improvements. (Although this comes as no surprise to us subjectivists)1) Their system lacks enough resolution to hear any differences. Most likely culprits are glare and grain. Usually both. (This is often due to the use of Pro Solid State amps like QSC, Crown and Crest. Although the use of zip cord and Rat Shack interconnects also plays a role here.)
2) Their aural perceptive abilities are weak to nonexistent, aka deafness. POLLYinFLA admits the Possibilities include the ingestion of too many hamburgers or putty ear syndrome. Plus a host of others, too many to mention. (I'd like to add they lack exposure to live unamplified music which is self evident, otherswise they'd know how bad their components sound.)
All of which leads to:
3) Jealousy. Envy. Faulty Logics. Probably from taking classes like static and dynamic. (I think bird-brained POLLYinFLA might be 100% correct here. I think jealousy plays a large role in the objectivist attitude. After all I can understand how owning cheap pro-solid state amps, zip cord speakerwires and Rat Shack interconnects could lead to the jealousy and envy that illuminates brightly through every post they make. Add that to thier faulty logic in believing science has already revealed everything that needs to be known as far as which measurements are critical for deteremining how the human ear/brain combo decides what does and doesn't sound like live unamplified music and one can easily see how and why these objectivists are so easily mislead.)Poor, poor POLLYinFLA he thinks he knows the truth when the reality is he hasn't the foggest idea of what's really happening. Hail King POLLYinFLA! His bird-brainedness thinks he has all the answers when the truth is he doesn't even know what the real question is...
Oh wait. I forgot. You gave up ranting.cheers,
AJ
The threshold for disproving something is higher than the threshold for saying it, which is a recipe for the accumulation of bullshit - Softky
Hi.They might be listening.
Talk real here, no "bullshit".
c-J
With pets you have to clean up real shyte, whereas here its simply flying out of mouths virtually.
Please keep it coming, the delusions of grandeur here eclipse anything that Comedy Central can churn out.cheers,
AJ
The threshold for disproving something is higher than the threshold for saying it, which is a recipe for the accumulation of bullshit - Softky
> Last month my audition between the CD 44.1KHz 16-bit & DVD-audio
> 196KHz 24-bit formats of the SAME identical sound tracks of two total
> different recordings (from two different labels), has proven to me the
> sonic difference is nite & day.How do you know they are the same?
The only sensible test is to take the high resolution signal and create the low resolution signal from it in a known manner. This is trivial to do and if you can hear night and day differences then I would suggest somebody is trying to fool you.
Hi.Of course I know. These are two-disc albums, one DVD-audio & one CD, mastered from the same master sound tracks, complete with all relevant certification from the production company. Are you suggesting the music/disc production firms tried to "fool" the consumers? This is a legal issue, bud.
If you don't possess the basic music instinct, change your hobby.
If your audio gears are so crappie failing to reproduce the sonic difference, replace them.If you want to get such 2-disc album to compare, ask me instead of
talking stupid.c-J
> These are two-disc albums, one DVD-audio & one CD, mastered from the
> same master sound tracks, complete with all relevant certification
> from the production company.Mastered from the same master sound tracks? Without stating they were mastered in the same way it strikes me as rather unwise to conclude the only difference between the two is the resolution and bit depth. I am not sure it is even implied. Is there something else that makes you think the processing of the two was the same?
I presume the certificate concerns the use of the original recorded source (which was what resolution by the way?) and not a certificate relating to the equivalence of the CD and DVD?
.
I am not interested in the discs, I am interested in why an audiophile is claiming to hear "night and day" differences between 44/16 and 192/24.Can I take your lack of response as an indication that you have now realised that whatever was leading you to think the two sources were the same does not hold?
"These are two-disc albums, one DVD-audio & one CD,"There is no guarantee that the provenance of both discs is the same, the only way to guarantee that is to decimate the 24/192 bit to 16/44.1 yourself and then compare. If you did not do that then your comparison is as useful as a dime that dropped down the drain.
And unsurprisingly, I side with Andy19191, if the decimation is decent, there will not be night and day differences between the two.
Music making the painting, recording it the photograph
Hi.An apple is never an orange.
So to compare a valid compare, you want to change the apple to an orange first before comparing, right?
Amazing !!!! Where is your logics?
c-J
If your goal is to compare the sonic differences between 16/44.1 and 24/192, then what I have suggest is a valid way of ensuring that the only the files being compared in the bit depth and sampling rate. That way you can be sure that the both apples originate from the same orchard :), sure beats presumption which is what you started out with.
Music making the painting, recording it the photograph
Hi.So what is your goal?
To enjoy music through "valid way of ensuring" hows & whys. Make sure you can still sleep well at nite if you can't find such valid ways.
I suggest you stick to 'just enjoying the music', as your comments stuff relating to audio technology has been pretty much useless.
Music making the painting, recording it the photograph
.
:)
Indeed. Hardware problems are certainly a possibility. Is he using "audiophile" hardware or normal hardware? Is he using different hardware for 192/24 and 44.1/16? But without knowing the relationship between the two sources it kind of makes knowing the rest somewhat irrelevant.
Hi.I use the same DVD-audio player to play both the DVD-audio & CD discs, all other gears remain identical with on-screen monitoring which digital format being played. NO guessing like you guys.
you do realize that your DVD player uses different lasers and different electronics in some of its signal path for the DVD than it uses for the CD playback, don't you?
Hi.We are talking about the end sonic results DVD-audio vs CD due to their different sample bits & sample frequencies as a consumer.
Are you telling us we need a special lab-built-calibrated disc player to play both DVD-audio & CD discs in their respective optimum conditions in a lab environment in order to validate such sonic results?
There is always different way to grow & handle an apply vs an orange. They both are not the same fruit.
Blu-ray Discs use total different laser beam (650nM) technology to achieve better video resoluion than DVD-video (being to 25GB, over 5 times of DVD-video capacity), yet the sonics still backs down to lowly CD liner PCM level. What a shame.
I really wish one day in the forseeable future, the Blu-ray video bigboys would be interested in the minute audiophiles market to cut Blu-ray audio discs with its huge Mb capacity.
No, I'm telling you that you are comparing more changes than just 44.1kHz/16bit to 96kHz/24bit. To compare, record both to the same medium (such as a DVD-A disk, which is not restricted to 24/96) and listen to both with same equipment, preferably without knowing first which is which.
Hi.Unless we are in the same business with all the instrumentation available to carry such tests precisely, you think we ordinary Joe Blows easily get recordings via the two different digital format
with consumer's type PC burners available to us?You THINK you can bank on the outcome of such homeblew disc burnings
to draw the conclusion? How are you so sure your homebrew burning comes out right?I said I already got discs of both 16/24bit 44.1/192KHz from the same music production labs to ready to compare from the same master sound tracks. What else can be more reliable & certain?
Professional lab production or your homebrew burning?
Hey, you're the ordinary Joe Blow who "got discs of both 16/24bit 44.1/192KHz from the same music production labs". So use your serious clout with the 'labs'! -- ask them to make disks that can actually be compared under equivalent conditions. On the same medium and, like suggested above, also with both made from the same recording, the 16 bit reduced from the 24. Your conclusion may be correct (or not), but the conditions of the comparison aren't enough, IMHO, to back it up.
.
No, I'm an Ordinary Joe without any mastering lab connections. I'm also a Joe who's not yet making any final judgements on whether 24/96 is a big improvement over 16/44.
> ...whether 24/96 is a big improvement over 16/44. <Depends on what you consider "big". It's not the difference between Radio Shack speakers and Maggie 20.1's but its more significant than, say, cable lifters. Bad example since I've never tried cable lifters... but anyway, the diffs are not huge but I consider them worthwhile.
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