• 06-22-2004, 02:52 PM
    Woochifer
    Quote:

    Originally Posted by WmAx
    It has not been demonstrated to be important. Their have been careful studies to attempt to confirm, but none that stood up to scrutiny have demonstrated positive results.

    If the 44.1/16 bandwidth is sufficient to cover all audible phenomena, then why is it that this bandwidth has never been the standard used by professional sound engineers and mixers with original recordings and during the mixing process?

    Quote:

    Originally Posted by WmAx
    As far as conclusion without proof? No. I CAN NOT conclude that your claim has any signfigance. Data does not suggest this conclusion. You would have me assume things are true before such has been proven?

    And I never suggested a conclusion. I'm pointing out that I've not ruled out any causal factors and have insufficient information to reach a conclusion. However you want to preclude variables without knowing anything about the source material is entirely your prerogative.
  • 06-22-2004, 03:05 PM
    kingdaddykeith
    Quote:

    Originally Posted by Woochifer
    If the 44.1/16 bandwidth is sufficient to cover all audible phenomena, then why is it that this bandwidth has never been the standard used by professional sound engineers and mixers with original recordings and during the mixing process? .

    It's always been my understanding that it is more of a headroom issue, the higher sampling rates allows for more headroom so the recording level can be hotter, which lowers the noise floor.
  • 06-22-2004, 03:09 PM
    WmAx
    Quote:

    If the 44.1/16 bandwidth is sufficient to cover all audible phenomena, then why is it that this bandwidth has never been the standard used by professional sound engineers and mixers with original recordings and during the mixing process?
    The optimal parameters for playback do not necsarrily translate into the optimal parameters for flexible/versatile attributes for recording/format change, etc.

    JAES, July/August 1978, Volume 26, Number 7/8, Page 562

    Here is the relevant quote:

    Quote:

    In the discussions of standards relative to digital audio to date we feel that the needs of broadcasting organizations have been little mentioned, and we would like to make a few points.
    In Europe a standard sampling rate of 32kHz +/- 50 parts per million, giving an audio bandwidth of 15kHz, has been agreed within the EBU for use by broadcasters. As commercial applications assume a bandwidth of about 20Khz, and hence sampling rates from 40-60 kHz, it is probable that broadcasters who will need to interface between these standards will do so by means of a digital rate-changing filter, so avoiding D/A and A/D conversion.
    To make this rate-changing filter as simple as possible to instrument, it is desirable to choose certain sampling frequencies for the commercial recording application. These in order of merit are:

    (1)_________48
    (2)40 ______________56
    (3) ___44 ________52 __60
    (4) __42 _46 ___50

    Each row of frequencies requires twice as many calculations in the filter as the previous one. For easy rate-changing of this kind, both the input and the output sampling rates should be locked, and so any choice of system-clock frequency should be integer related to 32kHz, as well as to the system sampling rate.
    -Chris
  • 06-22-2004, 03:17 PM
    WmAx
    Quote:

    Originally Posted by kingdaddykeith
    It's always been my understanding that it is more of a headroom issue, the higher sampling rates allows for more headroom so the recording level can be hotter, which lowers the noise floor.

    Higher bitrate, indeed, allows or eaiser digital recording. It allows a an additional safeguard against improper level settings to the recorder causing clipping(slight improper seetings won't be a disaster later on), clipping due to extraordinary dynamic sources, as well as allowing for different dithering processes to be used when reduced to standard 16 bit depth.

    -Chris
  • 06-22-2004, 03:26 PM
    Woochifer
    Quote:

    Originally Posted by Steve1000
    I'm certianly not going to buy into "hi-res" audio if it is inherently no better for two-channel music than CD "low-res" [??] audio. I won't buy into the new format simply because they are paying better attention to the mastering with the new format. A LOT of people join me in this sentiment. If this is what the recording companies are doing, "hi-res" is toast, IMHO.

    I have a VERY rudimentary understanding of these things. As I understand it, CDs are sampled at 44.1 khz, so that the frequency response maxes out at about 22 khz, which is well in excess of the hearing of the vast majority of the human population, though dogs may be able to appreciate it.

    I'm not going to be running double-blind of ABX tests between SACD and CD disks listening for audible consequences of 23 khz info in this lifetime. Life's too short, I'm not going to spend my money on such silliness if there's no support for it in theory, and I have too little expertise. If I am persuaded that CDs should have the same two-channel audio quailty as SACDs, DVD audio, etc., I'm not gonna bite for the "high-res" stuff, as a matter of principle. That' why I'm asking.

    The vast majority of households, including mine, have no interest whatsoever in anything more than highly euphonic two-channel sound or in trying to hear what little information is conveyed above 22 khz.

    I am quite willing to alter my views, but not based on the thin reed of purely subjective assertions.

    To me, it boils down to a very simple question. Do the high res discs improve upon the listening experience over what the CD versions offer?

    So far, whether it's the audible improvements I've observed with two-channel material, or with the whole new dimension of 5.1 surround mixes, my answer is a definite yes. The 96/24 discs I've bought thus far are a clear cut improvement upon their CD counterparts, and what 5.1 surround music brings to the table is a whole new way to enjoy music. If this is typical of SACD and DVD-A, then I see no drawback to it whatsoever.

    There are still plenty of poorly done CDs out there, and any chance to revisit these recordings and give them an improved transfer is welcome in my view. In addition, creating a 5.1 surround mix requires going back to the original multitrack master, which means that it's possible to obtain a higher resolution mixdown than a version that was originally done using analog recorders (and potentially degraded by going through successive iterations during the mixdown process on analog equipment). This would include the two-channel mixdown as well, if the artist chooses to have an album remixed at the higher resolution.

    You're more than welcome to quibble about what the true causal effect is, or choose not to get into high res based on some personal principle. I don't have the answer on what the true causal effects are (and absent access to the original source material, nobody else does either), and frankly, I don't care. In the meantime, I'll just enjoy what these new versions offer with better sound quality and listening to familiar music in a new way. In my view, results count and what I've observed so far, the new high res discs have delivered.
  • 06-22-2004, 03:43 PM
    Steve1000
    Fair enough! Your point of view is entirely reasonable, IMHO. Mine's just a little different. :)

    Quote:

    Originally Posted by Woochifer
    You're more than welcome to quibble about what the true causal effect is, or choose not to get into high res based on some personal principle. I don't have the answer on what the true causal effects are (and absent access to the original source material, nobody else does either), and frankly, I don't care. In the meantime, I'll just enjoy what these new versions offer with better sound quality and listening to familiar music in a new way. In my view, results count and what I've observed so far, the new high res discs have delivered.

  • 06-22-2004, 04:05 PM
    kingdaddykeith
    I'm starting to believe that a hi-Rez recording is better when down converted at the end of the chain. The DAD-A's and especially the SACDís that I've sampled are, well I hate to use the word grunge, but that what the higher frequencies sound like for the lack of a better word. Maybe itís some kind of artificial noise or something but it's not right to my ears. However when I buy a 24/96 DTS or DD 5.1 mix it sounds much more real and less fatiguing on the top and much fuller on the bottom then any of my SACDís.

    I donít know if this is true but I've read, and was told by a Parasound engineer that the higher the sampling rate the more noise, and the reason most all (if not all) 24/96 input compatible processors down convert to 16/48 (even my Halo C2) is because of the lack of technology to either filter or negate this noise. It makes sense at the recording end to have the most resolution and headroom as possible, but the playback of this full resolution has yet to impress me.
  • 06-23-2004, 06:46 AM
    Monstrous Mike
    Quote:

    Originally Posted by Woochifer
    If the 44.1/16 bandwidth is sufficient to cover all audible phenomena, then why is it that this bandwidth has never been the standard used by professional sound engineers and mixers with original recordings and during the mixing process?

    When working with digital audio it is better to record and process the audio at higher sampling rates and bit lengths and then produce the master CD at 44.1/16. If an audio engineer kept his signal at 44.1/16 from microphone to master, the possibility of some losses and distortions being introduced are more likely than when handling the digital audio at the higher levels during processing, mixing, filtering, normalization, equalization, etc.
  • 06-23-2004, 06:57 AM
    Monstrous Mike
    [QUOTE=WmAx]
    Quote:

    Sampling frequency(bandwidth) dictates the precison of sampling an amplitude in a given space of time. While in a simple analysis, the bandwidth of 20kHz is adequte to adress raw bandwidth sensitivy, in order to replicate the test tone circumstance times suggested(5us, etc.), a far higher bandwidth would be requrined in order to record/playback. For example, if y ou have an acoustic event that contains only audible data(<20kHz), this still does not account for the potential tiny time dealy differences between channels(ears) that will occur since each ear is a discrete sensor essentially. Apparently, the interchannel time sensitivity of human ears is far higher then the raw bandwidth detectability. For a simplistic model, imagine 2 microphones(let's pretend it has a 200khz bandwidth for this discussion) imagine a sound source, of the same spectral content that is within 20khz bandwidth, one mike is placed 2mm farther away then the other. Obviously, it woudl require a 170kHz bandwidth to accurately record and play back this difference betweeen the two sources. Your ears apparently have this type of effect. Just remember that this been demonstrated to be readily audible with special test tones, not music.

    -Chris
    I really don't have any idea what you are explaining here.

    The author of my previous quote is very clearly implying that a 96 kHz sampling rate is audibly better and a 192 kHz sampling rate is even better than that. And the implication is that the time distance between the samples can affect the imaging vis-a-vis a time delay.

    I agree that a phenomenon called "interaural time delay" can be detected by humans and is used in conjunction with intensity differences to determine a direction for sound. But like I said I do not see how time delay is related in any way to bit sampling rate.
  • 06-23-2004, 07:44 AM
    WmAx
    Quote:

    The author of my previous quote is very clearly implying that a 96 kHz sampling rate is audibly better and a 192 kHz sampling rate is even better than that. And the implication is that the time distance between the samples can affect the imaging vis-a-vis a time delay.
    I was not addressing the author, I was addressing the specific question/issue you presented about the interchannel time delays that he mentioned.

    Quote:

    I agree that a phenomenon called "interaural time delay" can be detected by humans and is used in conjunction with intensity differences to determine a direction for sound. But like I said I do not see how time delay is related in any way to bit sampling rate.
    I don't know what it has to do with bit sampling rate, either. BUt as for frequency sampling rate, how else would you propose to record/playback at a 5us or 1.5us accuracy with 44.1Khz? While the audible spectral informatin will be recorded with redbook format, the specific coordinates in time will be shifted into what can be stored in a 44.1kHz rate.

    Here is an over-simplified illustration of my understanding of this phenomena and how it related to sampling frequency:

    H=2mm(approx. 6us)(0dB)
    UUUUUUUU=17mm(50us)(1 cycle 20khz sine wave)

    Potential difference example:

    170Khz bandwidth limited
    L:
    HHHHHUUUUUUUUHHHUUUUUUUUHHHHHHHHHUUUUUUUUHH
    R:
    HHHHUUUUUUUUHHHHUUUUUUUUHHHHHHHHHHHUUUUUUUU

    20kHz bandwidth limited
    L:
    HHHHHHHHUUUUUUUU UUUUUUUUHHHHHHHHUUUUUUUU
    R:
    HHHHHHHHUUUUUUUU UUUUUUUUHHHHHHHHUUUUUUUU

    The higher bandiwdth can allow for the interchannel time difference to exist at finer resolution, as illustrated in the crude graphic above. While a 20kHz cycle is 50us in duration, the actual time at where this amplitude can actually originate is not limited. The 20kHz wavelength can begin at 200us or 204us or 201.3486 us, etc. Lower sampling rated reduces this possible difference relative the sampling rate limits.

    If my understanding is wrong, please explain.

    All of this and how it relates to audibility are a different issue.

    -Chris
  • 06-23-2004, 10:31 AM
    Monstrous Mike
    Quote:

    Originally Posted by WmAx
    I don't know what it has to do with bit sampling rate, either. BUt as for frequency sampling rate, how else would you propose to record/playback at a 5us or 1.5us accuracy with 44.1Khz?

    -Chris

    I do not know what you mean by a 5 microsecond accuracy.

    Let's look at this from another angle. Let's assume we have two signals which are identical but signal B is delayed by 5 microseconds. I presume this is what we are talking about. Now let's assume these signals are a 1kH sine wave with an amplitude of +/- 1 volt.

    And I presume the premise is that a 44.1 kHz sampling rate will not be able to accurate capture this time delay. Using 48 kHz (which is close to 44.1) the time between each sample is 20 microseconds. Again the premise is that this long period between samples is not sufficient to capture the 5 microsecond delay. I guess that would seem obvious at first glance.

    However we have our two identical 1 kHz sine waves where the second has started 5 microseconds behind the first. This represents a 1.8 degree phase shift. Therefore, sampling at 48 kHz (i.e. 20 microsecond intervals) yields this:

    Digital Sample Number One

    Time = 0 seconds
    Signal A = sin (0 degrees) = 0
    Signal B = sin (-1.8 degrees) = - 0.0314

    Digital Sample Number Two

    Time = 20 microseconds
    Signal A = sin (7.2 degrees) = 0.126
    Signal B = sin (5.4 degrees) = 0.0941

    Digital Sample Number Three

    Time = 40 microseconds
    Signal A = sin (14.4 degrees) = 0.249
    Signal B = sin (12.6 degrees) = 0.218
    Digital Sample Number Four

    Time = 60 microseconds
    Signal A = sin (21.6 degrees) = 0.377
    Signal B = sin (19.8 degrees) = 0.339
    .
    .

    Digital Sample Number Ten
    Time = 200 microseconds
    Signal A = sin (72.0 degrees) = 0.951
    Signal B = sin (70.2 degrees) = 0.941
    .
    .

    Digital Sample Number Thirty
    Time = 600 microseconds
    Signal A = sin (216.0 degrees) = - 0.588
    Signal B = sin (214.2 degrees) = - 0.562
    .
    . etc.

    I think that clearly shows that a 5 microsecond delay is captured quite well with a 20 microsecond sampling interval. As a matter of fact, I think the number of bits representing those analog amplitude values have more of an affect on capturing the delay than the sampling rate does.

    Further, using Nyquist's Theorum, the above will hold true for frequencies up to 24 kHz.
  • 06-23-2004, 11:35 AM
    WmAx
    Quote:

    However we have our two identical 1 kHz sine waves where the second has started 5 microseconds behind the first. This represents a 1.8 degree phase shift. Therefore, sampling at 48 kHz (i.e. 20 microsecond intervals) yields this:

    I think that clearly shows that a 5 microsecond delay is captured quite well with a 20 microsecond sampling interval.
    Thank you for the correction. I simulated this after your explanation, and you are correct, that the time difference is indeed more accurate then the raw sampling frequency lead me to believe. Appears I was shortsighted - thinking of only the sample frequency. My error.

    -Chris
  • 06-23-2004, 04:29 PM
    kingdaddykeith
    I though someone would find this interesting, itís a recording engineers opinion of the compression problem with CD's. Didnít read it all so forgive the poor description.

    http://georgegraham.com/compress.html
  • 06-23-2004, 04:31 PM
    Feanor
    Suppose the signal isn't a sine wave?
    Quote:

    Originally Posted by Monstrous Mike
    ...Let's look at this from another angle. Let's assume we have two signals which are identical but signal B is delayed by 5 microseconds. I presume this is what we are talking about. Now let's assume these signals are a 1kH sine wave with an amplitude of +/- 1 volt.

    ...
    I think that clearly shows that a 5 microsecond delay is captured quite well with a 20 microsecond sampling interval. As a matter of fact, I think the number of bits representing those analog amplitude values have more of an affect on capturing the delay than the sampling rate does.

    Further, using Nyquist's Theorum, the above will hold true for frequencies up to 24 kHz.

    Your example proves that 44.1KHz can distinguish two sine waves 5 us apart. But we don't listen to sine waves. Suppose there is a pair of complex wave forms where there are instantaneous spikes 5 us apart?
  • 06-23-2004, 05:22 PM
    WmAx
    Quote:

    Originally Posted by Feanor
    Your example proves that 44.1KHz can distinguish two sine waves 5 us apart. But we don't listen to sine waves. Suppose there is a pair of complex wave forms where there are instantaneous spikes 5 us apart?

    Spikes? You mean 'impulses'? That's not something you find in music or nature. However, with nearly any natural sound, you can express the signals as a sum of sine waves. This extends to any symmterical waveform(as opposed to assymetrical which is only common in a synthetic environment(usually test signals)): square wave, triangle wave, etc. They are a result of many sine waves.

    -Chris
  • 06-24-2004, 07:54 AM
    Feanor
    Impulses occur in "nature" ...
    Quote:

    Originally Posted by WmAx
    Spikes? You mean 'impulses'? That's not something you find in music or nature. ...

    ... if not in music. Isn't true that supersonic events, (at least), cause impulses? These everts aren't all that uncommon, e.g. base ball hit by a bat, crack of a bull wip, gun shots, some explosions. No wonder these things never sound real except heard live!
  • 06-24-2004, 08:18 AM
    Monstrous Mike
    Quote:

    Originally Posted by Feanor
    Your example proves that 44.1KHz can distinguish two sine waves 5 us apart. But we don't listen to sine waves. Suppose there is a pair of complex wave forms where there are instantaneous spikes 5 us apart?

    A spike of 5 us would certain be missed by a sampling interval of 20 us. However, that 5 us pulse would have frequency components in the 384 kHz range or greater and thus are not audible and cannot be reproduced by standard amplifiers or speakers. A sampling interval of 20 us represents a sampling rate of 48 kHz. According to Nyquist Theorem, that means it can only capture frequency components of 24 kHz or less. Spikes and other spurious noise over 24 kHz will not be captured.

    So a 48 kHz sampling rate can capture frequencies with a period of 20 us or more and can also capture two signals which are offset by 5 us (assuming a large enough bit word length).

    You are confusing time shifting with spectral content..
  • 06-24-2004, 09:19 AM
    WmAx
    Quote:

    Originally Posted by Feanor
    ... if not in music. Isn't true that supersonic events, (at least), cause impulses? These everts aren't all that uncommon, e.g. base ball hit by a bat, crack of a bull wip, gun shots, some explosions. No wonder these things never sound real except heard live!

    The events you describe are composed of primarily symmetrical waveforms. Perhaps not perfect, but I don't know. An impulse is assymetrical. However, this is immaterial. It is possible for some speakers to reproduce assymetrical waveforms with great accuracy.

    An assymetrical waveform is one that does not have inversely matching values in it's two 180 degree halves. These can be seperated and looked at as negative and positive sections of the waveform.

    Here are two simplified illustrations, represent a waveform with symetry and the same waveform without.

    Symmetrical Wave form

    + Pos
    .......H..............
    .....H...H...........
    ...H.......H......H.. 0 zero
    .............H....H..
    ................H.....
    - Neg


    Assymetrical(extreme - for illustration)

    + Pos
    ......H...............
    ....H..H.............
    ..H......HHH....... 0 zero
    ........................
    ........................
    -Neg

    -Chris
  • 06-24-2004, 04:00 PM
    Sir Terrence the Terrible
    Quote:

    All undeniably true if you remove the word audible from the end of this statement. You state this as if it's proven fact, and I am not aware of any research coming to this conclusion as far as audibility is concerned.
    Chris, it is not financially feasible for any engineer to sit around a wait for science to tell them what they already hear. It is well documented that engineers get better imaging from the use of higher sampling rates. It is well documented that engineers hear their mixes more clearly at higher sampling rates, so I don't think any intelligent engineer is going to sit around waiting for research on the issue.

    Quote:

    Certainly it improves accuracy. Audibly with normal music playback? I don't see subtantial evidence of this.
    What would constitute substantial to you? I mean considering that just about every studio in Los Angeles, New York, Memphis, and every other major city that has a large music community has migrated from 16/44.1khz to 24/96khz, I would call that VERY substantial. Someone had to have heard an audible improvement, or there would be nothing to justify the cost of the upgrade, which can run into hundreds of thousands of dollars. So if you are looking for science to prove what many already know, then by all means do so, but that doesn't make good business sense to me.

    Quote:

    That is the issue So far, the respected controlled tests/references on this subject have not been able to achieve a positive result.
    They have not been able to acheive a negative result either. So it would be short sighted to discount it altogether.

    Quote:

    did not state the contrary. I stated exactly this sediment, but I also stated that it is not logical to attribute the things 'credited' to hi-rez playback since their is no strong evidence that suggests that this should be the case. Until a peer-reviewed, scrutinized, valid audiblity test has been performed that achieves positive statistical signficance, then it can not be accepted as fact.
    I disagree with your perspective entirely. In case you didn't know it, I (like many other engineers) sit down for many hours testing and listening to new equipment to decide whether it is worth my investment. I (like many engineers) have my own set of test that allow me to do this in a way that I can make an educated decision. It is not my job to become a scientist, conduct listening test to obtain a statistical measure just to justify my purchase. That is inefficient and unnecessary. After I am finish testing a piece of equipment, I know for a fact that my decision to purchase, or not is an educated one. I do not need DBT , and a peer review to make that decision for me. It is my feeling that most engineers feel this way.

    (This is just my opinion) DBT, research and publishing for peer review is for the scientific community. That is not the job of a audio engineer. We only need one answer, does it sound better than my current equipment. According to polls taken at the Surround 2004 conference, about 86% of engineers polled believes that 24/96khz sounds better than 16/44.1khz. Is that scientific? No, but it leads me to believe that where there is smoke, there is fire.

    I have taken this position and I am going to pretty much stick by it for now. I have done my own homework listening to various recordings I have done at various bit and sample rates. I have used several recorders during the same session set at various bit and sample rates so I can play them back and listen. I made my decision based on what I heard. If I heard no differences between 44.1, 48, and 96khz, I would have probably stuck with 44.1 since it required no investment. That however was not the case, and I invested in what I thought sounded the best.

    Does the sample rate make a difference in sould quality? Definately. Why? I know it improves imaging, and the sound is cleaner and more distinct to the ear, but otherwise I don't know. Does bitrate matter? Only in recording and post production. I'll let the scientist figure the other crap out
  • 06-24-2004, 06:43 PM
    mtrycraft
    Chris, it is not financially feasible for any engineer to sit around a wait for science to tell them what they already hear.


    Or what they imagine to hear?
    They should at least see what science has to say about it when that data is available.


    It is well documented that engineers get better imaging from the use of higher sampling rates.

    What kind of documents? Not all documents are created equal.


    It is well documented that engineers hear their mixes more clearly at higher sampling rates,

    Same as above.

    o I don't think any intelligent engineer is going to sit around waiting for research on the issue.


    But what will that intelligent engineer do when the data is in? Or, cannot be demonstrated? Ignore it?


    I mean considering that just about every studio in Los Angeles, New York, Memphis, and every other major city that has a large music community has migrated from 16/44.1khz to 24/96khz, I would call that VERY substantial.


    Substantial only by numbers. Doesn't mean much beyond that though. After all a huge number of people on the planet believe in the supreme being.

    Someone had to have heard an audible improvement, or there would be nothing to justify the cost of the upgrade,


    That is absolute nonsense. One only has to look at the high end audio, and audio cable industry in specific.
    This is a trend driven by numerous drivers. Besides, mastering is different from consumer audio listening and reproduction.


    So if you are looking for science to prove what many already know,

    Or, what they only think they know as that is certainly not out of question and is certainly a valid and real possibiolity.





    In case you didn't know it, I (like many other engineers) sit down for many hours testing and listening to new equipment to decide whether it is worth my investment.


    Subjectively, of course, right? So, it is prone top bias and gullibility?

    It is not my job to become a scientist, conduct listening test to obtain a statistical measure just to justify my purchase.

    Ah, but if you did do such lisening tests, maybe you wouldn't follow the herd blindly and not waste you money foolishly?


    That is inefficient and unnecessary.

    Not if it gets you to an objective answer instead of guessing or just an expensive preference issue.

    After I am finish testing a piece of equipment, I know for a fact that my decision to purchase, or not is an educated one.

    How can you? It is based on a very subjective test prone to bias and unreliability.

    I do not need DBT ,

    That is unfortunate.



    It is my feeling that most engineers feel this way.

    That is unfortunate also.


    (This is just my opinion) DBT, research and publishing for peer review is for the scientific community.


    While you have this opinion, it is unfounded.

    That is not the job of a audio engineer.

    Why not? I would think you wanted real answers, the truths, not maybe or whatever.


    We only need one answer, does it sound better than my current equipment.


    That is the whole point. You don't know, not in an objective manner. You think you do but far from being a fact.

    According to polls taken at the Surround 2004 conference, about 86% of engineers polled believes that 24/96khz sounds better than 16/44.1khz. Is that scientific? No, but it leads me to believe that where there is smoke, there is fire.

    Well, at least you know it is not scientific. Why not find out for sure?
    A higher percent believe in the supreme being. Where there is smoke there is fire, right?
    How about psychics? Homeopathic medicines? We can go on and on, audio doesn't have immunity from nonsense, myths, hype, etc.
  • 06-24-2004, 06:48 PM
    Mr Peabody
    Chris, while attending a training session, many years ago, given by Harmon Kardon, they explained the benefit of wide bandwidth in amplifiers. At the time they were only one a few companies that had bandwidth ratings up to 100kHz. They said it was for harmonics. For example, when middle C is struck on a piano it vibrates the other strings up and down from it. Phono cartridges are capable of reproducing very high frequencies as well as good home speakers. It would seem if analog is a 100% signal that more sampling would bring you closer to representing that 100%. And maybe these harmonics are what people are missing when listening to digital.

    From what I understand of 5.1 music they do not use the extra channels for ambient reproduction but there is actually musical information in the rear channels. Is this true? If so, 2 channel may not be perfect but how can you fool yourself into thinking you are at a show with a guitar behind you in one channel and a trumpet in the other rear channel?

    Terry, it is interesting to hear you confess you use your ears to evaluate sound when you tell those on the HT forum their system is crap if they don't use a SPL meter and measure every little thing. What is even more interesting is that your buddy mtrycraft is trying to convince me that 3dB difference is barely audible at all.
  • 06-24-2004, 07:02 PM
    mtrycraft
    Quote:

    Originally Posted by Sir Terrence the Terrible

    Also read what Bob Stuart of Meridan Audio says about higher sampling rates.

    http://www.meridian-audio.com/w_paper/Coding2.PDF

    .


    Did you read the whole article by Stewart? Did you read how much emphasis he places on science and research, psychoacoustic data on hearing?

    Did you read page 8, 'Do we need more thatn 44.1 ...'

    I am shagrined to read the two references he offeres up by Ohashi as they are anything but credible and has since been shown to be flawed.

    Not much guessing on his part. He takes the science route, not what feels good.

    This paper was also presented at an AES conference as well and published as a preprint which I happen to have:)
  • 06-24-2004, 07:11 PM
    mtrycraft
    Quote:

    Originally Posted by Monstrous Mike
    However, for the life of me, I cannot see how the has anything to with the sampling frequency of the digital audio signal. Do you have any ideas?

    It doesn't. But, it has everything to do with the recording engineer setting up the left and right channels.

    This time difference between the two channels has an interesting impact:)
    Diana Deutsch has an interesting CD on this 'Musical Illusions and paradoxes' at Amazon.
  • 06-25-2004, 06:07 AM
    Monstrous Mike
    This audio business is a funny business indeed. To proceed with experimentation in audio engineering based on a presumption that humans can hear above 20 kHz is absurd, IMHO, and I'm sure will form the basis of the next generation of audio snake oil.

    I fully appreciate the need to digitally process (e.g. record, mix, master, filter, etc.) at word lengths greater than 16 bits and sampling rates greater than 44.1 kHz. But the PCM 16/44.1 signal has the dynamic range and frequency range that exceeds human hearing capabilities. Demonstrating that CDs in this format can sound bad is not proof that the format is incapable. It is proof that the digital processing was inadequate.

    The "more is better" attitude is not restricted to audio but it sure is prevalent in audio.
  • 06-25-2004, 01:42 PM
    Sir Terrence the Terrible
    Quote:

    Originally Posted by Terrence
    Chris, it is not financially feasible for any engineer to sit around a wait for science to tell them what they already hear.


    Quote:

    Originally Posted by Mtry
    Or what they imagine to hear?
    They should at least see what science has to say about it when that data is available.

    Mtry, sorry man, I do not play into the "imagined" stuff. IN THIS CASE if we go by what you say, then you are the only sane one, and 90% of the engineers are there are suffering from mass suggestion. That is not logical, and is quite arrogant on your behalf. I do not think people who make a living at listening to audio are that stupid. Either you hear a benefit of a higher sampling rate, or you don't. It is that simple.


    It is well documented that engineers get better imaging from the use of higher sampling rates.

    Quote:

    What kind of documents? Not all documents are created equal.
    Can you decode this response and play it back to me?


    It is well documented that engineers hear their mixes more clearly at higher sampling rates,

    Quote:

    Same as above.
    Yeah, same as above

    o I don't think any intelligent engineer is going to sit around waiting for research on the issue.


    Quote:

    But what will that intelligent engineer do when the data is in? Or, cannot be demonstrated? Ignore it?
    I guess the answer will come when we cross that bridge, right? We haven't gotten there yet.


    I mean considering that just about every studio in Los Angeles, New York, Memphis, and every other major city that has a large music community has migrated from 16/44.1khz to 24/96khz, I would call that VERY substantial.

    Quote:

    Substantial only by numbers. Doesn't mean much beyond that though. After all a huge number of people on the planet believe in the supreme being.
    What does a person believing in a supreme being have to do with audio? And what makes you think that a studio would invest hundreds of thousands of dollars on something that was a figment of their imagination. Your response is incredibly silly. I gather you don't think audio engineers are very smart, and are subject to hearing things. Incredible!


    Someone had to have heard an audible improvement, or there would be nothing to justify the cost of the upgrade,

    Quote:

    That is absolute nonsense. One only has to look at the high end audio, and audio cable industry in specific.
    This is a trend driven by numerous drivers. Besides, mastering is different from consumer audio listening and reproduction.
    Here is the problem with discussing recording with someone who has never done it. A piece of wire and a $8000.00 amp costs no where near a Sonic Solution DAW. These things cost $100,000-$200,000 , a far cry from a piece of wire, or any high end product. If there was no improvement in the sonics of this workstation, how could a studio(working on a margin)justify its costs? You are trying to use the woes of the high end audio and cable industry, and apply it to the recording industry. Sorry Mtry, this is a round peg, and you are trying to squeeze it into a square hole. Not the same.


    So if you are looking for science to prove what many already know,

    Quote:

    Or, what they only think they know as that is certainly not out of question and is certainly a valid and real possibiolity.
    What right do you think you have to question their judgement? Do you know more than they do? I do not think so, and everyone cannot be imagining everything. If left up to you everyone is delusional, and there is no reason to pursue any sonic improvements ever. That is not logical or reasonable, and VERY shortsighted.

    In case you didn't know it, I (like many other engineers) sit down for many hours testing and listening to new equipment to decide whether it is worth my investment.

    Quote:

    Subjectively, of course, right? So, it is prone top bias and gullibility?
    Audio quality is indeed a subjective thing don't you agree?. Some people like the sound of MP3, and I think it is crap. Some engineers(like myself) test randomly, and unlabeled do we do not know what is what. Some know exactly what they are listening to. The point is not to prove anything scientifically as YOU desire, but to listen and judge for yourself. Do you understand that concept, or are you too skeptical to actually LISTEN to music rather than testing it?

    It is not my job to become a scientist, conduct listening test to obtain a statistical measure just to justify my purchase.

    Quote:

    Ah, but if you did do such lisening tests, maybe you wouldn't follow the herd blindly and not waste you money foolishly?
    So that's what you think everyone is doing(except you of course), just being sheep. Mtry either you are the most airheaded individual in the world, or you are just plain arrogant as hell. EVERYONE is not blind and deaf as you loosely assert. Some people hear no difference between 48khz and 96khz sampling rate, and therefore remain stuck in redbook standards, and some hear a definate improvement and upgrade. I guess you would say that there is no audible improvement going from MP3 at 128kbps to 24/96khz

    That is inefficient and unnecessary.

    Quote:

    Not if it gets you to an objective answer instead of guessing or just an expensive preference issue.
    You are only assuming they are guessing, and that would be presumptuous on your part. No smart engineer or studio is going to invest hundreds of thousands of dollars in new equipment unless it has been rigorous tested by more than one individual(in the case of a studio) or objectively in the case of a smart freelancer. It would be too costly of a mistake for no benefit. Do you think you are the only one that thinks this stuff up?

    After I am finish testing a piece of equipment, I know for a fact that my decision to purchase, or not is an educated one.

    Quote:

    How can you? It is based on a very subjective test prone to bias and unreliability.
    How do you know what it is based on? I never released that information.

    I do not need DBT ,

    Quote:

    That is unfortunate.
    For you maybe.

    It is my feeling that most engineers feel this way.

    Quote:

    That is unfortunate also.
    Once again for you, not for us.


    (This is just my opinion) DBT, research and publishing for peer review is for the scientific community.

    [quote}While you have this opinion, it is unfounded.

    Who made you God so you could decide this?

    That is not the job of a audio engineer.

    Quote:

    Why not? I would think you wanted real answers, the truths, not maybe or whatever.
    You are assuming that an engineer testing methods do not yield accurate answers. More arrogance on your behalf. Maybe we are not quite as smart as you are in this area (sarcasm off)


    We only need one answer, does it sound better than my current equipment.


    Quote:

    That is the whole point. You don't know, not in an objective manner. You think you do but far from being a fact.
    Once again, how do you know YOU are correct? More presumptuous statements here

    According to polls taken at the Surround 2004 conference, about 86% of engineers polled believes that 24/96khz sounds better than 16/44.1khz. Is that scientific? No, but it leads me to believe that where there is smoke, there is fire.

    Quote:

    Well, at least you know it is not scientific. Why not find out for sure?
    A higher percent believe in the supreme being. Where there is smoke there is fire, right?
    How about psychics? Homeopathic medicines? We can go on and on, audio doesn't have immunity from nonsense, myths, hype, etc.
    Everything is hype to you. So why bother with anything? Audio may not be immune from nonsense, but everyone is not ignorant as you would believe either. So what is your approach, everyone is stupid until science proves them smart?

    Lets see, Mtry= no recordings, no experience recording, no recording education, but knows everything. Eliott Scheiner, Chuck Ainsley, Tony Brown, George Massenburg, Shawn Murphy and many more=almost a hundred years of experience between them, audio educated and degreed, thousands of recording between them, and they know nothing. Wow, Mtry you are a real legend(sarcasm off again)
  • 06-25-2004, 02:16 PM
    Sir Terrence the Terrible
    Quote:

    Originally Posted by Mr Peabody
    From what I understand of 5.1 music they do not use the extra channels for ambient reproduction but there is actually musical information in the rear channels. Is this true? If so, 2 channel may not be perfect but how can you fool yourself into thinking you are at a show with a guitar behind you in one channel and a trumpet in the other rear channel?

    Terry, it is interesting to hear you confess you use your ears to evaluate sound when you tell those on the HT forum their system is crap if they don't use a SPL meter and measure every little thing. What is even more interesting is that your buddy mtrycraft is trying to convince me that 3dB difference is barely audible at all.

    Mr Peabody;

    Your understanding is lacking quite a bit. The rear channels are used for both ambient information, and for musical information. It is the artist/producers choice how the mix is done, not yours. The answer to this question is it depends on the mix, as there are no hard fast rules for how to use the surround channels.

    How can you fool yourself into thinking that you are at a show when the audience is clapping behind the performers where two channel places them? Is that where they are in real life? I don't think so.

    Lastly, your reading comprehension is lacking. You can use your ears to measure the QUALITY of a signal. You CANNOT use your ears as MEASURING devices for amplitude. Your ears know what sounds good, but they cannot tell you that your speakers are precisely balanced with any accuracy. Room acoustics make this impossible, as do the fact that your ears do not know exactly how loud 75db is. How many recording have you said you have done?????

    A 3db imbalance is enough to pull the soundfield to the loudest channel, if you are talking about using 1khz as the test tone. 3db difference between channels in the bass region is very difficult to hear because of our hearing insensitivities in the bass region.

    A man with your profound recording background ought to be able to go toe to toe with Mtry effortlessly. So what's up?
  • 06-25-2004, 05:00 PM
    WmAx
    Quote:

    Originally Posted by Sir Terrence the Terrible
    Chris, it is not financially feasible for any engineer to sit around a wait for science to tell them what they already hear. It is well documented that engineers get better imaging from the use of higher sampling rates.....

    I mean considering that just about every studio in Los Angeles, New York, Memphis, and every other major city that has a large music community has migrated from 16/44.1khz to 24/96khz, I would call that VERY substantial. Someone had to have heard an audible improvement, or there would be nothing to justify the cost of the upgrade, which can run into hundreds of thousands of dollars. So if you are looking for science to prove what many already know, then by all means do so, but that doesn't make good business sense to me.....

    They have not been able to acheive a negative result either. So it would be short sighted to discount it altogether. ....

    I disagree with your perspective entirely.....

    This seems like a lot of explaining. This is not needed. I stated my position. I don't see any reason I should excuse a certain group of people from the requirements of proof that everyone else is bound to. I don't buy the 'popular opinion' - this is no substantial evidence. At one time, everyone believed the Earth was flat ... that the Earth was the center of the universe and other various fallacies. The opinion of hundreds of millions of people did not make it so.


    -Chris
  • 06-25-2004, 06:51 PM
    Woochifer
    T-man, Mtry, et al

    Just when I thought the general audio forum had gone peace and love on me, you guys decide to go old school! Now, I'm just waiting for references to space aliens, voodoo, and missile animations, and the cycle will be complete. You guys are keeping me young! :D
  • 06-25-2004, 08:48 PM
    mtrycraft
    Quote:

    Originally Posted by Woochifer
    T-man, Mtry, et al

    Just when I thought the general audio forum had gone peace and love on me, you guys decide to go old school! Now, I'm just waiting for references to space aliens, voodoo, and missile animations, and the cycle will be complete. You guys are keeping me young! :D


    Keeps me young too:)
  • 06-25-2004, 09:07 PM
    mtrycraft
    Quote:

    Originally Posted by Monstrous Mike
    This audio business is a funny business indeed. To proceed with experimentation in audio engineering based on a presumption that humans can hear above 20 kHz is absurd, IMHO, and I'm sure will form the basis of the next generation of audio snake oil.

    I fully appreciate the need to digitally process (e.g. record, mix, master, filter, etc.) at word lengths greater than 16 bits and sampling rates greater than 44.1 kHz. But the PCM 16/44.1 signal has the dynamic range and frequency range that exceeds human hearing capabilities. Demonstrating that CDs in this format can sound bad is not proof that the format is incapable. It is proof that the digital processing was inadequate.

    The "more is better" attitude is not restricted to audio but it sure is prevalent in audio.


    We know this but then one only needs to read SirT's responses. If most do it or believe it, it must be so, etc.
  • 06-25-2004, 09:15 PM
    mtrycraft
    Exhilarating
    exchange. So wonderful to get the facts from ones who know they are correct, never question anything or anyone.

    Oh, I cannot claim to be supernatural. That I will leave to others.
  • 06-26-2004, 07:33 AM
    DMK
    Quote:

    Originally Posted by mtrycraft
    Oh, I cannot claim to be supernatural. That I will leave to others.

    You may not be supernatural but I see you are an A/R Elite Member. There's just GOTTA be some perks in that! :D
  • 06-26-2004, 06:34 PM
    mtrycraft
    Quote:

    Originally Posted by DMK
    You may not be supernatural but I see you are an A/R Elite Member. There's just GOTTA be some perks in that! :D


    You bet:)
    More get to pound on me, longer :D
  • 06-28-2004, 02:02 PM
    Sir Terrence the Terrible
    Quote:

    Originally Posted by Monstrous Mike
    This audio business is a funny business indeed. To proceed with experimentation in audio engineering based on a presumption that humans can hear above 20 kHz is absurd, IMHO, and I'm sure will form the basis of the next generation of audio snake oil.

    I fully appreciate the need to digitally process (e.g. record, mix, master, filter, etc.) at word lengths greater than 16 bits and sampling rates greater than 44.1 kHz. But the PCM 16/44.1 signal has the dynamic range and frequency range that exceeds human hearing capabilities. Demonstrating that CDs in this format can sound bad is not proof that the format is incapable. It is proof that the digital processing was inadequate.

    The "more is better" attitude is not restricted to audio but it sure is prevalent in audio.

    MM,

    You have read enough of my posts to know that I do not believe in 75% of the things the music industry says. A scientist does and hearing experiment, publishes his results at AES, and the next year someone comes back to AES disputing that person findings. This merry go round goes on year after year.

    What you think is absurd may not be at all. At the last AES meeting, listening test were conducted to determine whether we could perceive tones, or overtones that lie outside of what is know to be the upper hearing of humans. This listening test proved inconclusive, with some individuals(who had great hearing tests) being able to hear the roll off of instruments with content above 20khz, and some not hearing a thing. So it maybe just a little quick to say this is absurd. While no one can say for sure, you certainly at this stage cannot rule ANY possibility out.

    I do not think any says that we can 'hear" above 20khz. I also do not think the switch to 96khz was about that. What I think (and what is echo'd by other's) is the accuracy of the higher sampling inband. . You can plainly see what 44.1khz does to a 1khz( I choose that frequency for my test) sinewave, and it's not pretty. When you see that same 1khz sinewave at 192khz sampling, it is spot on the original waveform. So you are right, in terms of dynamic range, and ability to produce signals up to the limit of human hearing, 16/44.1khz is adequate. If accurate tracking of the musical waveforms is highly desired, the 44.1khz just does not cut it.

    This same test was done at Surround 2002, except they used a signal much higher in frequency. Here is the link for your perusal.

    http://www.smr-home-theatre.org/surr.../page_07.shtml
  • 06-28-2004, 03:46 PM
    DcnBlu
    The Human Element
    I will keep this initial brief short, the idea that the human ear can hear the difference between 44.1Khz and 96.1Khz is a large stretch of the imagination. The average healthy human can hear 40hz to 18Khz without much concentration. Notes below ~40hz are not heard, they are felt, and notes above 18khz are heard by few. If you doubt me, go get a hearing test done at your local hospital, it will surprize you. How do I know you ask? I have over 15yrs in the medical profession to draw upon.

    Now when it comes to the electronic component of this little equation, lets keep in mind that all the pretty sine waves don't matter a bit, when it comes to "human" perception. Not one person alive can tell the difference in a 1khz wave at 44.1khz or at 192khz, sorry :eek:

    With this being said, I love a well put together system. The ability to be "transparent" in the audio path is a must. TRANSPARENT; performing its' given task without adding distortion or altering the signal. The proof in any system, IMO, is in its ability to reproduce the music as it was recorded, good or bad. I actually like finding/discovering poorly recorded material; shows my system is not bias... :p

    Again, let me go on the forum record for saying, MEDICALLY, it is impossible to hear any difference between a signal sampled at 44.1 or 96khz. The human ear doesn't even function in a way that would facilitate such discriminatory properties. :D A good reminder of this is the use of the terms Decibel and linear. Human hearing is linear, we notice a doubling of power when there is a 10db change in amplification. Audio equipment will actually double the power of a signal with a 3db change in amplification.
  • 06-28-2004, 04:10 PM
    Sir Terrence the Terrible
    Quote:

    Originally Posted by WmAx
    This seems like a lot of explaining. This is not needed. I stated my position. I don't see any reason I should excuse a certain group of people from the requirements of proof that everyone else is bound to. I don't buy the 'popular opinion' - this is no substantial evidence. At one time, everyone believed the Earth was flat ... that the Earth was the center of the universe and other various fallacies. The opinion of hundreds of millions of people did not make it so.


    -Chris

    Chris,

    I hate to be the one that bursts your bubble, but know one in the industry really cares about your position. They don't care whether you excuse them or not from providing proof. What you need to do if you are so gung ho about holding audio engineers to your personal scrutiny, is to visit a studio and listen for yourself. You can postulate and pontificate on this audio forum all you desire, but if you are looking for evidence that someone hears improvements at higher sampling rate, you will be waiting a long time. You have heard neither DVD-A or SACD, and already by technical means (and not actually listening) condemning them as unnecessary. Music is for listening, and that what engineers do, they listen. If you really don't believe anything, and EVERYTHING is a marketing ploy then take a trip to a studio, and listen for yourself. If you are firm in your position, and are not willing to take a trip to a studio and listen for yourself, then you will find yourself still here at audioreview complaining like mad that science is the only way to tell what your ears hear. Science cannot measure imaging. Science cannot measure hearing more tightness in percussion, or space and air around instruments. So to rely so purely on science in a medium that requires that you listen, can sometime make you walk away with half a picture.

    I am in the belief that more testing has to be done before anyone can discount anything. I think that bandwidth issue is pretty settled, but there are remaining issues yet to be explored. Rather than taking a hard fast position as you have, I will wait until more testing with filters, and converters and their influence on what we hear before I decide that 96khz is a waste. In the mean time, I will use what God gave me(my ears) and listening to those things that need listening to, and watch things that require my sight.

    As far as your analogy about the earth, its off base. Audio has already discovered the earth was round, now we are looking into what is akin to the oceans, or best said area's where was have not sonically explored yet.
  • 06-28-2004, 04:31 PM
    Sir Terrence the Terrible
    Quote:

    Originally Posted by DcnBlu
    I will keep this initial brief short, the idea that the human ear can hear the difference between 44.1Khz and 96.1Khz is a large stretch of the imagination.

    Since Chris is demanding scientific evidence to support everything under the sun, what scientific support do you have that backs your point?







    Quote:

    The average healthy human can hear 40hz to 18Khz without much concentration. Notes below ~40hz are not heard, they are felt, and notes above 18khz are heard by few.
    Not correct at all. Humans can hear 20hz signals, but below that they are felt. The average human cannot hear as high as 18khz, very few can. Lower that to around 16khz through your twenties(approximately) 16khz through part of your thirties, and no higher than 12khz above that. Those who have taken good care of the hearing do much better than those who have not.

    Quote:

    If you doubt me, go get a hearing test done at your local hospital, it will surprize you. How do I know you ask? I have over 15yrs in the medical profession to draw upon.
    I do every year. My last test showed I could hear signals up to 17khz, which is approximately 1khz down from last year.

    Quote:

    Now when it comes to the electronic component of this little equation, lets keep in mind that all the pretty sine waves don't matter a bit, when it comes to "human" perception. Not one person alive can tell the difference in a 1khz wave at 44.1khz or at 192khz, sorry :eek:
    So you are telling me that we will hear no difference in the most sensitive frequencies that we can hear? okay.........

    Quote:

    With this being said, I love a well put together system. The ability to be "transparent" in the audio path is a must. TRANSPARENT; performing its' given task without adding distortion or altering the signal.
    So what you are telling me so far is that 44.1khz sample rate is transparent compared to the analog original? okay........


    Quote:

    The proof in any system, IMO, is in its ability to reproduce the music as it was recorded, good or bad. I actually like finding/discovering poorly recorded material; shows my system is not bias... :p
    Do you realize music has gone through alot of transformation before it get's to your system? Compression, eq, conversion, downsampling etc. Unless you have spent hundreds of thousand of dollars, I seriously doubt your system is TRANSPARENT. Especially if the room is taken into consideration.

    Quote:

    Again, let me go on the forum record for saying, MEDICALLY, it is impossible to hear any difference between a signal sampled at 44.1 or 96khz. The human ear doesn't even function in a way that would facilitate such discriminatory properties. :D A good reminder of this is the use of the terms Decibel and linear. Human hearing is linear, we notice a doubling of power when there is a 10db change in amplification. Audio equipment will actually double the power of a signal with a 3db change in amplification.
    This is so off, I am not going to touch it. Human hearing linear????? How do you explain the fletcher-Munson curve if our hearing is linear. Wow!!!
  • 06-28-2004, 04:50 PM
    mtrycraft


    science is the only way to tell what your ears hear.


    That is the only way to tell in a reliable manner. If you don't believe that and it appears that you don't, that is unfortunate. I suppose acoustics was arrived without the scientific approach? Without DBT listeing?

    Science cannot measure imaging.

    You think? Perhaps phase shift between the two front speakers cannot be measured? But why measure it and spec it? As soon as you change your acoutic environment, it affects the phase shift at your ears.

    Science cannot measure hearing more tightness in percussion, or space and air around instruments.

    You really think this? I bet the scope can detect what you perceive that better than you can imagine it, at times.

    So to rely so purely on science in a medium that requires that you listen, can sometime make you walk away with half a picture.

    One thing to measure it which you stated cannot be done which would imply something beyond science and the physical world, kind of supernatural, and, another having a real need for it as these are greatly affected by the acoustic space, speakers and recording quality.
    Perhaps you deal in mysticism?


    I will use what God gave me(my ears) and listening to those things that need listening to, and watch things that require my sight.

    Unfortunately he also threw in a monkey wrench, the brain that can confuse the issue, make stuff up and fill in blank information or a different bit when nothing has changed. But, hey, blame science for that for discovering it. Throw it out. Burn them at the stake.
  • 06-28-2004, 06:55 PM
    DcnBlu
    Just for you Terrence
    Quote:

    Originally Posted by Sir Terrence the Terrible
    Since Chris is demanding scientific evidence to support everything under the sun, what scientific support do you have that backs your point?



    Not correct at all. Humans can hear 20hz signals, but below that they are felt. The average human cannot hear as high as 18khz, very few can. Lower that to around 16khz through your twenties(approximately) 16khz through part of your thirties, and no higher than 12khz above that. Those who have taken good care of the hearing do much better than those who have not.

    Yes I am correct, and I did not bother to break out the age chart to offer such data in this post. I did want to keep it short. First, you are wrong, young healthy adults can and do easily hear the range of 18 to 20k..I test them all the time. As with anything else some have better hearing than others. Those that earn an H-1 rating can hear into 18khz +. As we age we do experience hearing loss. Most of this loss comes from our neglect in protecting our ears during our "super human" youth (blasting loud music in the car, turning up the tone controls on our parents stereos ect).
    Since frequencies below 40hz or primarily non direcitonal, and very large ( a 44hz tone produces a wave larger than a mid size car) I am wondering where you get this information from. What you call hearing is actually a perceived feeling. If you want to get technical, yes our ears do respond to 20hz and 16hz and 13hz ect...the pressure wave at 90db is down right commanding. You can't "hear" them persay, you feel/perceive them.
    With that said, I think we must define hearing. That which is decernable and identifable from and to its source. But if you don't believe me, grab a telarc/bose testing CD that has 50hz, 40hz and 20hz tones played -10db. Please remember to use a SPL meter and begin the test at 50db and increase ect. ect.


    I do every year. My last test showed I could hear signals up to 17khz, which is approximately 1khz down from last year.

    I am glad you have your hearing tested yearly, the 1khz difference in the test means little to nothing. It could have been you had more wax in your ears that day than last year. There are any number of factors to explain the drop, even the fact that you don't hear as well. What was the SD from left to right for this test and last years test?

    So you are telling me that we will hear no difference in the most sensitive frequencies that we can hear? okay.........

    Not at all, and I never said that. Since that range covers several octaves (~800hz to 8khz) there will be peaks and vallies depending on where a person is most sensitive. Note, this will differ from person to person. On AVERAGE, this range is not the problem when it comes to hearing, and we all will hear it just fine. Damage to the TM or ET will alter our perception of these tones, so lets just stick with the healthy adult ages 20-50.

    So what you are telling me so far is that 44.1khz sample rate is transparent compared to the analog original? okay........

    Transparent = the ability to reproduce the analog signal without coloration. As far as our ears are concerned, the 44.1khz reproduction of the analog original is more than adequate. Without sine wave generators, our ears can't tell if the signal is a perfect copy of the original or not. Hence, to our ears 44.1khz makes a transparent copy. I would bet that even at lower sampling rates the average person couldn't tell the difference.


    Do you realize music has gone through alot of transformation before it get's to your system? Compression, eq, conversion, downsampling etc. Unless you have spent hundreds of thousand of dollars, I seriously doubt your system is TRANSPARENT. Especially if the room is taken into consideration.

    :) Yes, I am aware of the transformation music goes through before it gets to our system. These uncontrolable variables differ from company to company. Are you telling me you know just how each CD is made? Are stereo mics or dual mono mics used? My system is very transparent/neutral in its reporduction of the signal supplied on the original CD. That is where we take our measurements and such from, not the recording studio. If you do, wow, you must some awesome ties in the music world.

    This is so off, I am not going to touch it. Human hearing linear????? How do you explain the fletcher-Munson curve if our hearing is linear. Wow!!!

    Lets discuss this. I know the fletcher-Munson test. Did you even review the sampling data used?
    http://www.webervst.com/fm1.gif

    This is a copy of the graphed results. Notice not one person could hear the test tones in the 20hz range. Looks more like 40ish to me. On the high end 20khz was not reached either, maybe 18k. Look at the intensity of the signal used to even the lowest freq range, somewhere 100db (average loudness was kept to around 85db for testing purposes) to hear the lowest noted freq. Now this is only a graph of the end result, but lets use it as a basis to end this nick-picking on every little variable...unless you just want to ;) I am enjoying this so. Oh did you also notice that they raised the intensity of each tone in 3db steps (linear raising the power/doubling it). LOL, I am sorry I noticed I said linear hearing, ack. I am incorrect, we hear analog, machines are linear :D. I see why you gave me the mental frown. You could have just pointed to my mistake though. If you notice any other obvious errors please just say so, I'm typing this while watching cartoons with my son. Oh, spelling errors don't count, that's my wife's expertise. Either way this test does support my original statement; we hear (~40hz to around 18khz) on average.

    If the original poster wants loads of research data, more power to him. I am not going to spend hours typing a single point of view, audio is a "living" hobby, it grows as the equipment becomes better and the users become more demanding on reaching that near point in dopple-ganging the original signal flawlessly.

    The whole point of digital is to mimic analog. Happy listening everyone.
  • 06-28-2004, 07:35 PM
    WmAx
    Quote:

    Originally Posted by Sir Terrence the Terrible
    Chris,

    I hate to be the one that bursts your bubble, but know one in the industry really cares about your position. They don't care whether you excuse them or not from providing proof.

    I do not see any new issues being brought up. Only a defense of non-scientific evaluation. You can refer to my previous replies, as if i reply at this point, it will be redundant.

    -Chris