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Ok, yet another dumb digital question...As my DAC understands signals up to 24/96, is it possible to buy software to convert standard WAV files into a higher resolution??

Only 2 channel or multichannel.

Do you want to do this on the fly, or re-store all of it as96kHz sampling?

At what resolution?

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Ok, yet another dumb digital question...As my DAC understands signals up to 24/96, is it possible to buy software to convert standard WAV files into a higher resolution??

Only 2 channel or multichannel.

Do you want to do this on the fly, or re-store all of it as96kHz sampling?

At what resolution?

Lossless on the fly when exporting the files to an iPOD, it an iPOD will accept such a thing...

any Daw will.

like

www.audacity.sourceforge.net

or
http://www.reaper.fm/

but please recognize that there is no benefit whatsoever to upsampling a "redbook" resolution source for the purpose of playback as you cannot put in what was never there to begin with..

there are advantages for such a thing if you were to be doing any processing(adding effects, eq, crossfades, etc) but for playback? leave it at redbook..otherwise you are taking up more space for nothing.

Well, Music Giants has some high resolution tracks.

a
B&W music club is free for 3 months..EP's on 44.1/24bit....Others:

HD tracks is one of the better ones/Linn also..

http://www.musicgiants.com

http://www.itrax.com

http://www.highdeftapetransfers.com

http://www.linnrecords.com

http://www.dgmlive.com

http://www.hdtracks.com

www2.deutschegrammophon.com

music.e-onkyo.com/contents/hd.asp

http://www.classicrecords.com

http://www.2l.no/hires/index.html

http://www.shockwave-sound.com

http://www.referencerecordings.com

http://www.magnatune.com

http://www.unipheyemusic.com

http://boomkat.com

http://dancemusichub.com

http://www.junodownload.com

http://www.stompy.com

http://www.trackitdown.net

http://www.turntablelab.com/digital

http://www.acousence.de/index.html

http://www.hifitrack.com/en

http://www.gimell.com/

http://www.archive.org/details/etree

the above is free(archive.org)

you can do a search for 24 bit files, genre, artist, even the recording equipment used. fantastic live stuff there.

the world is your oyster, JimV

dont forget about bt.etree.org and dimeadozen...

im looking at dime right now...tons of 24 bit files..all legal and free..

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Lossless on the fly when exporting the files to an iPOD, it an iPOD will accept such a thing...

There's no such thing as "lossless" in upsampling, even though you don't have to remove any content, it is (and I mean this literally and mathematically) impossible not to add some minor rolloff or ripple in the frequency domain. It is possible to keep all artifacts under 16 bits, or 24 bits, however.

In real time? That's a lot of FLOPS.

Audacity has a resampler. I say this with some hesitation, since I haven't tested the resampler inside of it.

i repeat. no benefits.

http://www.mlssa.com/pdf/Upsampling-theory-rev-2.pdf

http://www.thetadigital.com/upsampling.htm

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In your OPINION only.

It's only his opinion?

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If you have peer reviewed papers that show there are no benefits, then please produce or point us to them. Showing us other people's theories or trying to debate your OPINION as fact is a waste of time.

Of course, the paragraph quoted above is profoundly dishonest, since the person you're responding to is simply reporting the well-understood scientific understanding of the time. You don't find many papers on "rocks fall down", either, now do you. Are you going to tell me "rocks fall up" because I can't produce a modern, peer-reviewed paper that says rocks fall down.

I will add one caveat: There is no difference AS LONG AS YOUR EQUIPMENT WORKS CORRECTLY.

Bear in mind that gain changes, etc, that go along with changes in sample rate, by themselves, can trivially create both an illusions and/or actual audible differences that are not due to upsampling.

thanks, JJ.

Mr. Lavry's sampling theory paper is a good read.

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

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thanks, JJ.

No problem. One of the joys of being the mainstream is that in fact there is not a lot of current peer-reviewed work on the veriest basics like the sampling theorem, reconstruction and reconstruction error, upsampling, time resolution of PCM, etc, because IT IS WELL UNDERSTOOD IN THE ACTUAL MAINSTREAM.

For instance, the Nyquest theory that Dan mentions, which is actually proved as theorem as part of the Shannon Information Theory paper (no, that's not its name, that's what the paper founded).

Some correlaries are the issues about time resolution (you get better time resolution with a reconstruction filter than you get without, the fact that if you couldn't get subsample time resolution your cell phone wouldn't work, etc... This is all tied together in one paper of Claude Shannon's. If people demand a reference, that would be the reference. If you ask any information theorist about "Shannon's Paper" they'll know where to send you. I don't have the cite at my fingertips, so I won't cite it right now.

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Mr. Lavry's sampling theory paper is a good read.

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

It's a pretty good white paper. Some coverage of delta-sigma would round it out. The tutorial at www.aes.org/sections/pnw/ppt.htm would round out the delta-sigma (or sigma-delta, depends on if you're Jim Candy and Joe Condon, or Dan Weiss and Bob Cordell....) aspects.

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but please recognize that there is no benefit whatsoever to upsampling a "redbook" resolution source for the purpose of playback as you cannot put in what was never there to begin with..

Not necessarily true. Granted there's no additional resolution. But as everyone knows, the final sonic results are more than about just resolution/sampling rate of the bitstream.

There are many dependent factors including the performance of the DAC with 24/96k stream versus 16/44.1k stream, the design of the output filter, etc.

I wouldn't be so quick to make a blanket statement of "no benefit whatsoever".

Too bad we have to deal with 16/44 to begin with. Remember these gents also believed that 16/44 was perfect and no higher needed, even calling us "intellectual frauds".

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This paper concludes that the slow roll-off characteristic of upsampling digital filters is indeed responsible for the improved sound quality....

Imagine using a higher format with slower filter to begin.
But we have to settle for less. Although J_J insinuates there is only one side of the story, mainstream science says otherwise. In fact, Dr. Kunchur, multiple national organizations, expert university professors that Dr. Kunchur visited, anonymous referees etc all disagree with J_J.

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The main reason is that the digital audio data residing on a CD is already irreparably time smeared. No amount post-processing of the digital audio data by the playback system can possibly remove or reduce this time smearing.

So 16/44 and sharp filters cause degradation as Dr. Kunchur's work, multiple national organizations, over a dozen anonymous referees believe otherwise. But I guess we have to settle for less.

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Therefore, the slow roll-off filter has better time resolution, or equivalently, less time smearing.

So higher than 16/44 is desireable with slow roll-off filters.

It is a shame, Lecteur, that we have to settle for less than the best to begin with. We should not have had too.
But then that is what these so called "mainstreamers", believe. Of course on other strings, we found these "mainstreamers", J_J etc, disagreeing with multiple PhDs at universities, national organizations, anonymous referees, and companies, such as Sony etc.
But as one can see, the evidence once again says otherwise?

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So 16/44 and sharp filters cause degradation. But J_J is dead set against anything higher,

The statement quoted above is completely false, and can be seen to be false by anyone who has the dilligence to dig through the tutorial slides that I've posted links for many times here.

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So 16/44 and sharp filters cause degradation. But J_J is dead set against anything higher,

The statement quoted above is completely false, and can be seen to be false by anyone who has the dilligence to dig through the tutorial slides that I've posted links for many times here.

I edited my post while J_J was responding, but I stand on the quote.

I see you could only quoted part of the post, interesting. Also it's what you have stated here, your position here that is the question. Are you now reversing your position and stating that you now agree with Dr. Kunchur, multiple mainstream national organizations, over a dozen referees, mainstream university professors, and Sony and other manufacturers that higher sampling/resolution of music is better than 16/44 reproduction? That is quite a reversal in position J_J and congradulations, if it is true.

But as we have seen many times before, you change positions more often than a politician. Are you going to be posting your change in position, and apologizing after 50 some pages of attacks on Dr. Kunchur's work, higher sampling/resolution necessary than 16/44, in the Dr. Kunchur strings, where it counts.

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Are you going to be posting your change in position,

My position is unchanged, and you've never once acknowleged what it actually is.

I don't think you know. You've been so busy spewing s**t about what it is, claiming to know for a fact, but you don't even know what my position is.

As to Kuncher... Kuncher made a couple of trivially visible, elementary mistakes regarding PCM systems, and you're going to have to live with that, too. Notice how his allegations about noise levels vs. resolution just disappeared one day, for instance? Notice how he begged the meaning of "resolve".

Now, stop harrassing me, and admit that you have never, ever found a single error in my postings on that subject.

And then leave. You've s**t in your own backyard here, fella, and you've picked a fight with someone who is
1) absolutely right
2) unwilling to kowtow to your nonsense

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Are you going to be posting your change in position,

My position is unchanged, and you've never once acknowleged what it actually is.

I don't think you know. You've been so busy spewing s**t about what it is, claiming to know for a fact, but you don't even know what my position is.

As to Kuncher... Kuncher made a couple of trivially visible, elementary mistakes regarding PCM systems, and you're going to have to live with that, too. Notice how his allegations about noise levels vs. resolution just disappeared one day, for instance? Notice how he begged the meaning of "resolve".

Now, stop harrassing me, and admit that you have never, ever found a single error in my postings on that subject.

And then leave. You've s**t in your own backyard here, fella, and you've picked a fight with someone who is
1) absolutely right
2) unwilling to kowtow to your nonsense

Hmmmmm. Dr. Kunchur concluded that 16/44 is not high enough, and you claimed otherwise; arguing for 50 pages plus, and for weeks, that Dr. Kunchur, and others were wrong and 16/44 is all that is needed, not any higher. Now you claim you never argued such a position.

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Hmmmmm. Dr. Kunchur concluded that 16/44 is not high enough, and you claimed otherwise;

One more lie from you.

That is not what I said.

Please stop lying.

Has everybody completely missed the point about the ipod?
The ipod part makes the whole question invalid.

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Has everybody completely missed the point about the ipod?
The ipod part makes the whole question invalid.

no it doesnt.

ALAC/itunes.. supports 24 bit. as does aforementioned ROCKBOX

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Hmmmmm. Dr. Kunchur concluded that 16/44 is not high enough, and you claimed otherwise;

One more lie from you.

That is not what I said.

Please stop lying.

Interesting reply to say the least as J_J has continually demonstrated he dislikes anything above approx 20khz, about 16/44.

J_J/Woodenville:

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Of course a 44.1kHz/16 bit system can resolve time to well under 5 microseconds. So something has gone wrong and as quoted, without surrounding context, the quote looks fundamentally ignorant of the basics of sampling theory.

Interesting since replicating ultra sonic frequencies faithfully requires reproducing ultra sonic frequencies faithfully, which each channel of a 44khz cd player won't do with a 20khz analog bandwidth. We are not talking pulsing two channels or other deviate trickery. We really need ultra-sonic frequencies in each channel to faithfully reproduce all the music. Dr. Kunchur as well as multiple major national organizations and corporations such as Sony, Hitachi, Philips etc agree. Even NCdrawal's reference strongly suggests using higher than 16/44 to begin with.

In Dr. Kunchur's conclusion:

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indicating that digital sampling rates used in consumer audio are insufficient for fully preserving transparency.

JJ

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The problem remains: Nothing quoted to the author, NOTHING WHATSOEVER, supports the contention of ultrasonic signals being required. NOTHING.

As we just saw, J_J is incorrect in his personal opinion.

Dr. Kunchur, 3 mainstream organizations, professors of other universities, over a dozen anonymous referees agree with Dr. Kunchur's 5 year study/test and conclusion that higher rates for ultrasonic frequency reproduction are necessary. Lest we forget major manufacturers like Sony, Hitachi, Philips etc have designed and manufacture such.

J_J:

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Give it up. You have no evidence that ultrasonic signals are required, GIVEN THE QUOTES YOU HAVE CITED. That's how it is. Live with it. Accept it.

Again J_J's personal opinion is against mainstream science listed above. So my statement stands.

As we have seen above, J_J's opinion is not shared by real mainstream science. This next quote is J_Js attempt to reproduce two signals with 10us shift using two sub-ultrasonic channels. However, Dr. Kunchur's conclusion is based on each channel handling ultra sonic frequencies. Unfortunately J_J does this for a whopping 28 pages.
Obviously J_Js grossly misleading test is designed to undermine Dr. Kunchur.

JJ

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If you do that experiment at 44.1 with 10 microsecond shift, and compare simultaneous with one channel shifted, you'll hear subtle, but reliable shift in sound in headphones.

When the two channel was shown to be the gross distortion it was, then the change to one external reference and one channel, another manipulated and gross distortion.

Dr. Kunchur himself:

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Many misconceptions and mysteries surround the perception and reproduction of musical sounds. Specifications such as frequency response and certain common distortions provide an inadequate indication of the sound quality, whereas accuracy in the time domain is known to significantly influence audio transparency. While the upper frequency cutoff of human hearing is around 18 kHz (or even lower in older individuals) a much higher bandwidth and temporal resolution can influence the perception of sound.

Non-linearities and temporal complexities in the auditory system negate the simple f ~ 1/t reciprocal relationship between frequency and time. In our group's research -- which lies at the intersection of psychophysics, human hearing, and high-end audio -- we measure the limits of human hearing and relate them to the neurophysiology of the auditory system. These experiments also help to define the criteria for perfect fidelity in a sound-reproduction system.

Our recent behavioral studies on human subjects proved that humans can discern timing alterations on a 5 microsecond time scale, indicating that digital sampling rates used in consumer audio are insufficient for fully preserving transparency.

Dr. Kunchur emailed me, upon request, the rigours of his 5 year study through publishing.

Dr. Kunchur:

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For those who have no idea what science and the (incredibly rigorous) scientific process is, let me explain what went into publishing the two above mentioned papers that have apparently generated controversy among lay readers (but no controversy whatsoever in all the professional circles, which include audiolists, otolaryngologists, acousticians, engineers, and physicists ).

An experiment has to be carefully thought out and then submitted as a proposal to an Institutional Review Board (IRB) and approved by them before it can even begin. Then optimum equipment, methods, and a multitude of cross checks has to be developed (if you read my papers in their entirety, you will appreciated what went in).

The results, analysis, and conclusions are then carefully considered and discussed with colleagues who are experts in their related interdiscplinary fields; for this I went in person to various universities and research institutes and met with people in departments of physics, engineering, psychology, neuroscience, music, communications sciences, physiology, and materials science.

After that the results and conclusions were presented at conferences of the

Acoustical Society of America (ASA),

Association of Research in Otolaryngology (ARO), and

American Physical Society (APS).

Seminars were also made at numerous universities and research/industrial institutions (please see the list on my web site). After each presentation, the audience is free to tear apart the conclusions and ask all possible questions. Eminent people such as presidents of the above mentioned societies and corporations (ASA, ARO, Bose corporation, etc.) have been present during my presentations.

After passing through this grueling oral presentation process, written manuscripts were then submitted to journals. There, anonymous referees are free to attack the submission in any way they want. More than a dozen referees and editors have been involved in this journal refereeing process.

Only after everyone is satisfied with the accuracy of the results and all statements made in the manuscript, are the papers published in the journals. The entire process took around 5 years from initial concept to refereed publications.

As one can see Dr. Kunchur's work is real mainstream science so where does that leave J_Js radical and misguided opinions.

Next J_J changes from two channels, to one external reference and one channel, again grossly misleading the consumer to "prove" his opinion.

J_J:

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But, for your information, given an absolute outside reference, yes, you can detect a 5 microsecond shift in a pulse in ONE channel.

Any test but a single channel test.

Incorrect and misleading science does not undermine Dr. Kunchur and his work, nor mainstream national organizations, referees, independent major manufacturers etc.

I think we should all stick to 3rd party, real mainstream science; not some cleverly disguised, left field, manipulation techniques.

it was me who found the paper,(from Kunchur) made contact with Kunchur initially, Sammett, though you enjoy taking credit, and thats cool... at any rate,

there are no advantages to upconverting redbook to high res for playback. none. whatsoever.

if one were working within a DAW environment, doing processing/crossfades/eq/etc , then yes, sure.. higher res files are of some use..but taking a low rate and upconverting for playback?

that is... to put it plainly..worthless...not only that, it eats up disk space.

you won't win the JJ Argument, Sammet. Stay in your lane.

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it was me who found the paper,(from Kunchur) made contact with Kunchur initially, Sammett, though you enjoy taking credit, and thats cool... at any rate,

That is because I did email Dr. Kunchur and received the above reply, 5-27-2009. We also communicated at other times.

It is also interesting that both of you have vigiously backed up your close friend here who has admitted faking data/graph for years, so viewers would sabotage their own audio systems and increase his sales. So all three of you working a confidence game Ncdrawal?

http://en.wikipedia.org/wiki/Confidence_trick

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A confidence trick or confidence game (also known as a bunko, con, flim flam, gaffle, grift, hustle, scam, scheme, or swindle) is an attempt to defraud a person or group by gaining their confidence. The victim is known as the mark, and any accomplices are known as shills.

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It is also interesting that both of you have vigiously backed up your close friend here who has admitted faking data/graph for years

Who brought Ethan up?? Talking about Ethan Winer is inappropriate...if for no other reason than it is way, way , way off topic.Go grind your axe somewhere else. Your trash talking has gotten old.. way past the point of being funny. You do realize that you come off as unprofessional, what with your whining about Ethan every other post, yes? Grow up. Find another obsession. You sound like a little kid that has had his lunch money taken away.

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Has everybody completely missed the point about the ipod?
The ipod part makes the whole question invalid.

You have recaptured my interest....Why exactly? The thing is a 160G iPOD classic. It accepts WAV files and I can bypass the onboard DAC with the Wadia and go straight to my DAC...

I have listened to folk tell me over and over that part quality has zip to do with computer based sound as bits is bits...well all I do is take those bits and launder them through an iPod theoretically without change in that bits as bits stream or file...

What do I have wrong...why is a WAV file on an iPOD any less useful than the identical WAV file coming from a computer??

the IPOD is fine, depending on how you get the bits to it..

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Has everybody completely missed the point about the ipod?
The ipod part makes the whole question invalid.

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You have recaptured my interest....Why exactly?

iPods won't work with anything higher than 48/16.