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24bit vs 16bit, the myth exploded! - Page 110

post #1636 of 1923
Quote:
Originally Posted by groovyd View Post
 

in theory 48khz allows for a 24khz bandwidth of which they say humans only have about 20khz range of hearing.  so the extra 4k would be for good measure but the reality is all of this makes a number of assumptions that are not true in a real system.

 

recognizing nyquist theory is one thing but if you think for a second about how that sampled waveform will look you see that 2 samples per cycle is really not enough even at a minimum.  take a 1hz sine wave and sample it 2 times per cycle.  it is entirely possible that both of your samples fall at EXACTLY the two zero crossings that occur in every cycle once on the way up and again on the way down and your reconstructed output becomes a flat line. by this it becomes pretty clear to see that 2 samples per cycle really is not enough at all to reconstruct your sine wave.  nyquist theory assumes a number of ideal conditions, none of which hold in a real sampling system.  it merely represents an absolute bottom line minimum spec requirement if you want to represent ANY of the information at a certain frequency but not necessarily all of it in all circumstances.  infact given 2 samples per wave taken at any point in the cycle i can draw for you an infinite number of possible near sine waves of different amplitudes that pass through those points.  lets assume the amplitude of the signal matters.  by raising the sample rate beyond the nyquist rate you quickly begin to remove the singularities and the multitude of waveforms that fit the samples for a given frequency and you allow breathing room for the realities of the limitations of a real sampling system, such as non-ideal pre-ADC bandpass and post-DAC reconstruction filtering.

 

if we assume 20khz is a reasonable limitation of human hearing without regard for the fact that our amp, speakers, etc further attenuate the response at these edge of spec frequencies

then 48khz is really still an absolute minimum. in my opinion 96khz gives plenty of breathing room to counter all of the additive realities of a true system from end to end. 

Watch this video.  I am beginning to think it should be required viewing before one can post about digital audio.

 

http://www.xiph.org/video/vid2.shtml

 

Not singling you out personally, but your post contains several fallacies which are repeated millions of times.  Watch and understand this 23 minute video, and you will get why they are false.  Not false just in theory, but in actuality.  Two or more samples is enough to fully reconstruct the signal, and one and only one waveform fits any possible combination of samples as long as no frequencies exceed half the sample rate.  Yes, filters are imperfect, and reconstruction falls just a bit short of perfect theory.  But most the of the important factors have been dealt with.  We can get something like 95% or more of what is predicted by theory and put the last few percent of inaccuracies in a place where humans do not hear them.  Effectively, audibly very, very close to 100% fidelity to humans in actual use. 

 

You can go to 96 khz if you just really, really, really want to be sure.  There certainly seems close to zero point, and beyond 96khz there is zero point.  It has never been shown in a credible repeatable test that people hear 96 khz vs 48 or even 44 khz.  Null results so far.


Edited by esldude - 6/10/14 at 10:30am
post #1637 of 1923

Super audible frequencies are completely superfluous. They add absolutely nothing to the sound quality of music. You might as well stuff your file full of excelsior. It serves no purpose to human hearing. In fact, even the upper audible frequencies are pretty unimportant. A lot of music doesn't contain those frequencies. You can't perceive them as musical notes. A good chunk of the population can't hear all the way up to 20kHz anyway.

 

It's just more kicking the line along a bit further just to satisfy OCD. Redbook is fully capable of reproducing sound with everything humans can hear.

post #1638 of 1923
Quote:
Originally Posted by esldude View Post
 

 Two or more samples is enough to fully reconstruct the signal, and one and only one waveform fits any possible combination of samples as long as no frequencies exceed half the sample rate.

 

This is a technicality, from theoretical point of view it is impossible to recover a sinosoidal with frequency of precisely half the sampling frequency. It gets folded into DC.

 

On a different note, you shouldn't believe everything you hear on the internet. I personally have tendencies in trusting my university professor who's published numerous papers on IEEE over some confident kid with a video camera and a computer.

post #1639 of 1923

I prefer well conducted listening tests over either of those!

post #1640 of 1923
Quote:
Originally Posted by bigshot View Post
 

Super audible frequencies are completely superfluous. They add absolutely nothing to the sound quality of music. You might as well stuff your file full of excelsior. It serves no purpose to human hearing. In fact, even the upper audible frequencies are pretty unimportant. A lot of music doesn't contain those frequencies. You can't perceive them as musical notes. A good chunk of the population can't hear all the way up to 20kHz anyway.

 

It's just more kicking the line along a bit further just to satisfy OCD. Redbook is fully capable of reproducing sound with everything humans can hear.

 

I'm sure this has been discussed countless times already so this is bound to lead nowhere. As we all know truncated frequency spectrum leads to slight amounts distortion to the time-domain waveform (at least if you consider things in context of Nyquist sampling). I am sure we all can agree that there is plenty of energy beyond 20KHz.

 

Regardless of whether it can or can't be heard directly by vast majority of the population in nominally noisy environments, I argue there is no longer an overwhelming reason not to minimize the amount of distortion introduced by modern reproduction systems.

post #1641 of 1923
Quote:
Originally Posted by bigshot View Post
 

I prefer well conducted listening tests over either of those!

 

Theory provides us with a foundation upon which we can build. It's a means to close the loop - to understand the problem top to bottom.

Pure theory has no value if it can't be translated into a tangible thing. A tangible thing has no weight if it is unexplainable through a common foundation. Without a satisfactory basis, all discussion becomes merely crossfire of opinions.

post #1642 of 1923

I'm putting together a stereo system to listen to music. The ultimate goal is to listen to music. Theory is great for helping to solve problems, when problems exist. But for the purposes of listening to music, if I can't hear it, it isn't a problem. Armchair Einsteins can spend all their time thinking about sound they can't hear. I would rather spend that time listening to music.

post #1643 of 1923
Quote:
Originally Posted by Digitalchkn View Post
 

I'm sure this has been discussed countless times already so this is bound to lead nowhere. As we all know truncated frequency spectrum leads to slight amounts distortion to the time-domain waveform (at least if you consider things in context of Nyquist sampling). I am sure we all can agree that there is plenty of energy beyond 20KHz.

 

Listening tests have shown that super audible frequencies add absolutely nothing to the sound quality of recorded music, even when a person can perceive the frequencies as sound pressure in test tones. In fact, there was one test where people didn't even think that sound quality of music was greatly affected by rolling the entire top octave of audible sound off. There is not plenty of energy beyond 20kHz with acoustic instruments. That's beyond any fundamental and beyond the important harmonics. Even if we could hear beyond 20kHz, those upper frequencies would probably be inaudible anyway due to masking by much louder lower fundamentals and harmonics.

post #1644 of 1923
Quote:
Originally Posted by bigshot View Post
 

 

There is not plenty of energy beyond 20kHz with acoustic instruments.

 

I have spectrum analyzed lots of commercial "hi-rez" classical and jazz and saw that modern systems can pick up spectral content to no less than 30KHz. Even older recordings made during the infancy of magnetic medium have a positive SNR beyond 20KHz.

 

Basically if you take your redbook audio  recording of your acoustic instrument and your 22.05KHz frequency point is wiggling in your spectrum analyzer then there bound to be energy beyond 22.05KHz. It's a different debate whether you or I are able to hear a test tone up there or not.

 

As an engineer, I ask myself how much deterministic energy is up there and if it's sufficiently large then I would want to capture it.  Oddly enough, I have yet to see some  analysis tool that plots integrated energy(power) over frequency as it applies to audio applications. That would allow one to determine 90 or 95% bandwidth requirements.


Edited by Digitalchkn - 6/10/14 at 11:44am
post #1645 of 1923
Quote:
Originally Posted by Digitalchkn View Post
 

 

Basically if you take your redbook audio  recording of your acoustic instrument and your 22.05KHz frequency point is wiggling in your spectrum analyzer then there bound to be energy beyond 22.05KHz. It's a different debate whether you or I are able to hear a test tone up there or not.

 

The stuff I can hear is what matters to me. I'm not putting together a stereo system for my dogs and the bats in the attic to enjoy. Besides, the dogs don't dig Brubeck. (I haven't asked the bats.)

post #1646 of 1923
Quote:
Originally Posted by bigshot View Post
 

 

The stuff I can hear is what matters to me. I'm not putting together a stereo system for my dogs and the bats in the attic to enjoy.

 

Sure. That's why, as a consumer, you make a choice. Your choice may differ from choice of others who may find benefit from a, say, 24/48 setup. However, your choice need not be universally accepted by all. I feel it is important for others to do their homework prior to making this choice.  I think this forum should be a place for all to learn and understand the basics of what's involved and then make an educated choice based on this information. Assertive messages to the effect of "I can't tell a difference therefore you shouldn't either" is of not much help to anyone.

post #1647 of 1923
Quote:
Originally Posted by bigshot View Post
 

 

The stuff I can hear is what matters to me. I'm not putting together a stereo system for my dogs and the bats in the attic to enjoy. Besides, the dogs don't dig Brubeck. (I haven't asked the bats.)

 

You'd need to get above 20khz.  :wink:

post #1648 of 1923
Quote:
Originally Posted by Digitalchkn View Post
 

 

I have spectrum analyzed lots of commercial "hi-rez" classical and jazz and saw that modern systems can pick up spectral content to no less than 30KHz. Even older recordings made during the infancy of magnetic medium have a positive SNR beyond 20KHz.

 

Basically if you take your redbook audio  recording of your acoustic instrument and your 22.05KHz frequency point is wiggling in your spectrum analyzer then there bound to be energy beyond 22.05KHz. It's a different debate whether you or I are able to hear a test tone up there or not.

 

As an engineer, I ask myself how much deterministic energy is up there and if it's sufficiently large then I would want to capture it.  Oddly enough, I have yet to see some  analysis tool that plots integrated energy(power) over frequency as it applies to audio applications. That would allow one to determine 90 or 95% bandwidth requirements.

 

The Balinese Gamelan has "musical" energy up to about 50 KHz and a cymbal crash can have energy above noise up to 102 KHz with 40% of it above 20 KHz (according to James Boyk - http://www.cco.caltech.edu/~boyk/spectra/spectra.htm)  if played absurdly loudly, to make sure you do not lose this would necessitate sampling at a rate in excess of 200 KHz - a bit like making sure your digicam captures gamma rays ;)

post #1649 of 1923
Quote:
Originally Posted by nick_charles View Post
 

 

The Balinese Gamelan has "musical" energy up to about 50 KHz and a cymbal crash can have energy above noise up to 102 KHz with 40% of it above 20 KHz (according to James Boyk - http://www.cco.caltech.edu/~boyk/spectra/spectra.htm)  if played absurdly loudly, to make sure you do not lose this would necessitate sampling at a rate in excess of 200 KHz - a bit like making sure your digicam captures gamma rays ;)


Hey, If there is a supernova when i'm taking a scenic photograph, I sure as heck want to capture all those xrays, gamma rays, neutrinos, and high energy particles that come screaming in from space. I hear they really add to the sparkle and shine in some of the landscapes panoramas.

 

Cheers

post #1650 of 1923
Quote:
Originally Posted by Digitalchkn View Post
 

 

This is a technicality, from theoretical point of view it is impossible to recover a sinosoidal with frequency of precisely half the sampling frequency. It gets folded into DC.

 

On a different note, you shouldn't believe everything you hear on the internet. I personally have tendencies in trusting my university professor who's published numerous papers on IEEE over some confident kid with a video camera and a computer.


Okay, so with perfect filtering you can fully reconstruct 22,049 hz, but not 22,050.  Glad you pointed it out as that hertz would have been terrible to lose.  Lack of theoretically perfect filtering is why we have a transition band.  Full response to 20 khz and close enough to full rejection not to matter at 22.05 khz.  The transition band can have some things going on, but that is where putting the artifacts of less than theoretical perfection is a place we aren't going to hear. 

 

As for calling Monty Montgomery some confident kid with a vidcam and computer, one always must tar the messenger when one has no reasoning to refute.   Of course what do you do when the widely published in IEEE prof also agrees with the fellow on the video?

 

One reason the video is instructive even to those who don't know, understand or care to be bothered by the theory is how it is all checked with quality equipment in the analog realm.  Signals are generated with very high quality, low distortion analog sources, undergo AD then DA conversion and checked again with all analog low distortion scopes and frequency analyzers.  Most of the most common claims about the inabilities or distortions of digital audio are shown to not be so at all. If you can use analog at both ends and show these myths of digital inadequacy for what they are...fallacious imagination it wouldn't matter if the video was by a kindergartner or a 100 year old Nobel prize winning professor.  Myths are debunked either way. 


Edited by esldude - 6/10/14 at 1:50pm
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