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Can someone explain audio resolution to me? - Page 4

post #46 of 77

Sorry, I didn't see this reply earlier. I appreciate the detailed response!

Quote:
Originally Posted by jaddie View Post

"Yeah, that's the now infamous "Human Hearing Beats the Fourier Uncertainty Principle" paper.  The title is mostly what's wrong with it, it's quite misleading and sensational.  The paper sheds no new light, however, and contrary to what the title seems to imply, the paper does not "prove" that humans can hear things that can't be measured.  The guys a hydrogenaudio have pretty much cleared this up already."

 

As in other references here and on other forums, I personally don't consider forum comments as authoritative in refuting peer-reviewed journal-published empirical research. No doubt there's plenty to learn on forums (I'm learning a lot here), but it's just a matter of how research is conducted and vetted. I very much appreciate the AES's publication record of ongoing research in this area. I agree about the title (journal editors LOVE a catchy title)--after reading the paper, I would have thought something long and awkward like "This Aspect of Human Hearing Is Not Subject to the Constraints of the Fourier Uncertainty Principle When Considered In This Context".

 

"As I think about how to respond, I've decided to simply ask what you mean by "time domain of sound", and how you're applying it to D/A processing, resulting in "time-shifting and blurring". This is mainly a reaction to the comments which always pop up first in discussions of digital audio sample rates, which revolve around the inaudibility of frequencies beyond 22 kHz (for me personally, more like beyond 18 kHz, looks like). But the digital signals don't just represent frequencies--they arrive at particular times, and artifacts like pre- and post-ringing can change the signal-onset point in time. And of course, multiple signals arrive at the ears at different times, subject to very fine discrimination by human hearing. In re Stereo, what I'm wondering about is how accurately different sound playback chains replicate the stereo timing of the source signals. References always welcome, of course.

 

On "temporal resolution cues", I freely admit as always that my own little foobar tests have no consequences for the science of psychoacoustics nor audio engineering. However the foobar results I've seen were actually obtained, I can say that I always fail to get anywhere in the neighborhood of significant results when I try to listen for frequency content: timbre changes, tonal changes, EQ differences of any kind. I do obtain significant results when I'm listening to the reverb, the room ambience, the apparent soundstage size and depth--features which depend at least in part upon when sounds arrive, as opposed to just their perceived timbre when they do arrive.

 

 

"I would advise extreme caution when drawing conclusions based on auditioning up-sampled files against their original.  True ABX testing is very difficult to do because of hardware limitations.  It's also impossible to isolate the effects of up-sampling from the effects of DACs being asked to perform an entirely different task in terms of filtering, the specific up-sampling algorithm, etc.  Up-sampling doesn't add any information, of course, so to tune in on what the "difference" is, we'd need to know a whole lot about the DAC, how it operates, what changes with a rate change, and most important, accomplish a DBT.  That would take two identical sets of hardware, playback sync (the hard part), and a true ABX comparator.  If we don't do the test that way, the observations are so polluted with bias that there can be no reliable conclusion."

 

I agree completely, and in detail. My own subjective experience has no consequences for the science--as you say, no reliable conclusions may be drawn. And in fact, I do suspect, without having anywhere near the equipment or circumstances to follow through, that "the effects of up-sampling from the effects of DACs being asked to perform an entirely different task in terms of filtering, the specific up-sampling algorithm" are precisely where the differences I experience do actually lie. My interest is primarily practical, and limited to my specific listening circumstances and tastes: I have some CD's; they sound better to me when upsampled; I have no meaningful reason NOT to do it.


Edited by UltMusicSnob - 8/21/13 at 8:26pm
post #47 of 77
Perhaps your redbook playback is broken somehow. Could it be that your CD player isn't performing to spec?
post #48 of 77
Quote:
Originally Posted by bigshot View Post

Perhaps your redbook playback is broken somehow. Could it be that your CD player isn't performing to spec?


That's possible, but I should have clarified. I paid a $$$ fortune $$$ to import these CD's from Japan (I have researched the P2P download networks, but I always pay legitimate <sigh> prices for real CD's), so I'm completely paranoid about scratching them.

They play in the CD player just once, to rip to Hard Drive, and then are put away. 

So *all* my listening is done from .wav files off the hard drive through my studio monitors.

 

Also, I've gotten excellent foobar ABX confidence levels on two completely different platforms, so any flaw that would be causing a difference would have to be duplicated on two totally unrelated playback chains. Possible, but less likely. I intend to try a third and fourth, especially if I can get hold of another interface with quality comparable to the RME Babyface.


Edited by UltMusicSnob - 8/21/13 at 9:51pm
post #49 of 77

@UltMusicSnob,

 

Can you give a estimate on how well you are able to (say with your eyes closed) identify the direction of a sound source? If the source only makes low frequency sound (say less than 300 Hz)? if the source only makes high frequency sound (say more than 4000 Hz)?

 

Cheers

 

edit: also, can you remind us what the details of your replay chain(s) are? sound player, dsp, OS, soundcard/dac, amplifier?


Edited by ab initio - 8/22/13 at 2:57am
post #50 of 77

I for one think Montgomery's video at xiph.org did an excellent job describing the temporal resolution of a bandlimited signal. It shows exactly how does the bandlimited signal and it's timing work with the samples and the filtering upon D/A conversion. I found this statement quite vague, what do you exactly mean with the below sentence?

 

Quote:
Originally Posted by UltMusicSnob View Post

Montgomery's video is a great presentation, it just doesn't go to the full range of questions about temporal resolution which are available to consider and study.

post #51 of 77
Quote:
Originally Posted by UltMusicSnob View Post
 
Hmm. Analog, yes, simple not so much. It's a stereo signal with a complex waveform, decoded by mental processes into multiple spatial components, subject to many stages of processing between original creation and final listening. To me, it's far from simple. I *don't* mean mysterious or unknowable. I *do* mean complex and not fully understood.

I think in this case you are making something seem much more complicated and mystical than it really is. Up until the point that they are being processed in the brain, every channel of an audio recording is a simple function--- an amplitude as a function of time. If you have a stereo or binaural recording, then you simply have two sets of audio signals that you play back simultaneously, one in a left speaker and the other in the right speaker. However, those are completely separate up until that point of play back.

 

An audio signal is a real-valued, bandwidth limited signal. Yes, if you made a graph of it, it certainly can look like it squiggles around a lot, but that doesn't change the fact that it's simply a real-valued function of time. How your brain processes audio information is irrelevant to the fact that audio signals are byte time histories in a computer, groove displacement time histories on a record, magnetic field time histories on a tape, voltage time histories in your audio amplifier, or pressure time histories in the air.

 

Anybody who understands that each channel of audio can be represented as a continuous amplitude function of time fully understands audio signals. It's the physics involved with recording transducers, storage media, playback electroincs, and playback transducers that aren't 100% perfect, but that doesn't change the fact that the limitations are well understood.

 

 

Cheers

post #52 of 77
Quote:
Originally Posted by UltMusicSnob View Post

 

As in other references here and on other forums, I personally don't consider forum comments as authoritative in refuting peer-reviewed journal-published empirical research.

 

'Authoritative' doesn't even come into it; a peer-reviewed journal article can be flawed, and this is independent of whether the flaw is described in Nature or written on the back of a napkin.

post #53 of 77
Quote:
Originally Posted by ab initio View Post

@UltMusicSnob,

 

Can you give a estimate on how well you are able to (say with your eyes closed) identify the direction of a sound source? If the source only makes low frequency sound (say less than 300 Hz)? if the source only makes high frequency sound (say more than 4000 Hz)?

 

Cheers

 

edit: also, can you remind us what the details of your replay chain(s) are? sound player, dsp, OS, soundcard/dac, amplifier?


I know as a matter of acoustics that the low frequency sounds are much less directional, the high frequency ones more so; thus the widespread use of single woofers and separated satellite tweeters. I don't know what my personal limits are, that's a good question. Since directionality is derived from higher freq's and my own hearing does not extend past 18 kHz, I would guess that I can discriminate less well than a healthy 18-year-old, but at least well enough to match a healthy 50-year-old.

This is the replay chain in order:

  .wav file on hard drive

  Lenovo desktop PC (don't know the motherboard make, but we're still in digital domain at this point)

  foobar2000 player with ABX utility

  external RME Babyface audio interface

  analog line out of Babyface to Schiit Asgard2 headphone amplifier

  Beyerdynamic DT 770 Pro headphones. - I have some Beyerdynamic DT 48's still, I ought to try those.

 

This playback chain is constant all the way back to the .wav files. Redbook is played back from a .wav file created by ripping the original CD to 44.1 / 16 in SoundForge 10, using iZotope 64-bit SRC, with 32 db filter steepness.

post #54 of 77
Quote:
Originally Posted by Cat Face View Post

I for one think Montgomery's video at xiph.org did an excellent job describing the temporal resolution of a bandlimited signal. It shows exactly how does the bandlimited signal and it's timing work with the samples and the filtering upon D/A conversion. I found this statement quite vague, what do you exactly mean with the below sentence?

 

Montgomery's video is a great presentation, it just doesn't go to the full range of questions about temporal resolution which are available to consider and study.

The dial on Monty's machine can move the waveform back and forth at the freq and phase he's chosen; that's great. The question for quality is whether or not the waveform proceeding from the final stage of the playback chain replicates not only the frequency content, but the timing, in two channels of a stereo track, of the live source that was initially captured [clearly, it does not do so with perfect precision, the question is how, where, to what extent, changes occur].

post #55 of 77
Quote:
Originally Posted by ab initio View Post

I think in this case you are making something seem much more complicated and mystical than it really is. Up until the point that they are being processed in the brain, every channel of an audio recording is a simple function--- an amplitude as a function of time. If you have a stereo or binaural recording, then you simply have two sets of audio signals that you play back simultaneously, one in a left speaker and the other in the right speaker. However, those are completely separate up until that point of play back.

 

An audio signal is a real-valued, bandwidth limited signal. Yes, if you made a graph of it, it certainly can look like it squiggles around a lot, but that doesn't change the fact that it's simply a real-valued function of time. How your brain processes audio information is irrelevant to the fact that audio signals are byte time histories in a computer, groove displacement time histories on a record, magnetic field time histories on a tape, voltage time histories in your audio amplifier, or pressure time histories in the air.

 

Anybody who understands that each channel of audio can be represented as a continuous amplitude function of time fully understands audio signals. It's the physics involved with recording transducers, storage media, playback electroincs, and playback transducers that aren't 100% perfect, but that doesn't change the fact that the limitations are well understood.

 

 

Cheers


To the stages you list between data and brain above, I would add the transmission stages which occur between pressure time histories in the air and the auditory processing neurons within the brain. I'm also thinking of the effects of processing on the shape of the wave--thus all the variable parameters in my iZotope SRC, for example, and the tradeoffs between filter steepness and other desirable characteristics. And of course, both A/D and D/A processes create differences from the original input source, as do amplification stages between microphone and A/D, and between Line Out and speakers.

 

I think we're saying the same thing, sort of. I'm not referring to anything bizarre and hidden about waveforms, just about the fallibility of getting the final output waveform in a high-quality reproduction of the input.

post #56 of 77
Quote:
Originally Posted by higbvuyb View Post

 

'Authoritative' doesn't even come into it; a peer-reviewed journal article can be flawed, and this is independent of whether the flaw is described in Nature or written on the back of a napkin.


Okay, I'm not wedded to the word "authoritative". I do notice that peer-reviewed journal articles are examined under double-blind conditions--just like recommended audio ABX testing. The logical reasons for doing so are quite similar in both cases. I also expect refutations of empirical research to present empirical research--not just comments on a forum.

post #57 of 77

there is a large body of practical theory - taught from textbooks in engineering courses - used by engineers, scientists daily in designing, building equipment, instrumentation, experimental apparatus, measuring, analyzing digital analog data

 

it is possible to criticize a paper when instrumentation, methods, commentary don't agree with this body of knowledge or reflect even "good enough" selection of physical, analytic tools - I mean, really, Grados?

 

Kunchur has made comments in his papers, in online discussions showing ignorance of EE "Signal and Systems", DSP course material, practical application to even PC Audio DAC available at the time of his experiments - much less the capability of industrial DAQ a physicist should be familiar with

 

and as a Physicist whose professional subject is superconductivity Kunchur is at best a serious amateur in Psychoacoustics

post #58 of 77
Quote:
Originally Posted by jcx View Post

there is a large body of practical theory - taught from textbooks in engineering courses - used by engineers, scientists daily in designing, building equipment, instrumentation, experimental apparatus, measuring, analyzing digital analog data

 

it is possible to criticize a paper when instrumentation, methods, commentary don't agree with this body of knowledge or reflect even "good enough" selection of physical, analytic tools - I mean, really, Grados?

 

Kunchur has made comments in his papers, in online discussions showing ignorance of EE "Signal and Systems", DSP course material, practical application to even PC Audio DAC available at the time of his experiments - much less the capability of industrial DAQ a physicist should be familiar with

 

and as a Physicist whose professional subject is superconductivity Kunchur is at best a serious amateur in Psychoacoustics


Any citations on these topics are welcome.

post #59 of 77
Quote:
Originally Posted by Cat Face View Post

I for one think Montgomery's video at xiph.org did an excellent job describing the temporal resolution of a bandlimited signal. It shows exactly how does the bandlimited signal and it's timing work with the samples and the filtering upon D/A conversion.

Yes, that video certainly does prove the point! I even handed it over on a silver platter when I listed the proof as starting at 21:55 into the video. Go to that spot, then watch as the spikes slide along smoothly at times that fall between the samples. If that's not absolute and total proof, I don't know what else to say!

As for the debunking at Hydrogen Audio, I think it's important for people not just to read stuff, but also to try to comprehend it. Admittedly, for those of us who don't follow advanced math, someone's explanation in a forum post might seem like an opinion, even when it clearly is in fact proof. There's also this from Post #52 above:

'Authoritative' doesn't even come into it; a peer-reviewed journal article can be flawed, and this is independent of whether the flaw is described in Nature or written on the back of a napkin.

--Ethan
post #60 of 77
Quote:
Originally Posted by EthanWiner View Post


Yes, that video certainly does prove the point! I even handed it over on a silver platter when I listed the proof as starting at 21:55 into the video. Go to that spot, then watch as the spikes slide along smoothly at times that fall between the samples. If that's not absolute and total proof, I don't know what else to say!

As for the debunking at Hydrogen Audio, I think it's important for people not just to read stuff, but also to try to comprehend it. Admittedly, for those of us who don't follow advanced math, someone's explanation in a forum post might seem like an opinion, even when it clearly is in fact proof. There's also this from Post #52 above:

'Authoritative' doesn't even come into it; a peer-reviewed journal article can be flawed, and this is independent of whether the flaw is described in Nature or written on the back of a napkin.

--Ethan


I understand that you and many others are satisfied with the information provided by the video--that's fine. I don't have anything to add to what I've already posted, so I'm going to leave it at that, I'm not trying to provoke anyone. I do wish you'd take my word for it that I have watched that video before, more than once.

I consider the question of high resolution audio research to be ongoing, and I will follow its further developments with interest.

Thank you for all the tips, references, and information, it's much appreciated.

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