No difference between 24/48 and 24/96!

Discussion in 'Mixing & Song Critique' started by Berry, Mar 6, 2005.

  1. Berry

    Berry Guest

    I was just doing my everyday browse through the web and nearly fell asleep until I found this pretty nice dissertation by two graduates at Detmold University over here in Germany which actually confirmed what I have always been thinking or at least been very suspicious about the industry's hype!

    Unfortunately, the dissertation is in German, but for those who speak and understand it, here is the link:

    It basically says that there was no big difference between 24/48 and 24/96 compared to the original analog signal. The two 48kHz converters they tested even bet the two 96KHz ones. Besides technical measurement, there was an audition. Most people chose the 48 signal for being the closest to the original. A couple of people even considered the 48 being the original. The two graduates claim that the quality of converters depends on a high SNR and a good analog circuit design. Note that this is just a very short and hence unprecise extract from the graduates' dissertation.
  2. TeddyG

    TeddyG Well-Known Member

    Jan 20, 2005
    It is my understanding that higher rates, after a certain point(Professionals I know say 24/48 is that point?), are, indeed, inaudible in any human hearing test. No surprise here, we cannot "hear" 48k, let alone 96. We also cannot "hear" any "faults" at 16/44.1(Or less). So, why don't we just use 44.1?(Or less)

    The reason one uses higher rates is supposed to be that starting with the highest possible rate - assuming all equipment is completely capable of retaining all quality - assures a higher level of quality --- later...

    I cannot speak of digital rate quality, I don't have the proper equipment, or the ears.

    I can speak of analog quality, as regards, say, reel-to-reel audio tape recording.

    At speeds at least as low as 1-7/8 ips(Cassette machine quality), "original" recordings can sound pretty "fine" - again, assumning good quality equipment. But if one tries to "copy" such tapes, quality diminishes quickly. After several generations(Copy to copy to copy, etc.), which might be done, for instance, if a cassette tape was a "master" recording, to add compression, or EQ or reverb, to the copy that was not on the master, then to, say, make a "final" copy for duplication(Any number of copies may be done, each to "add" it's own changes), quality really suffers.

    On the other hand, if one records the original master at, say 30ips(Rare, but done), then keeps each succeeding "generation" in the process at 30ips, then makes a "final" master tape, at say, 15ips, for duplication, with the ultimate goal of producing 1-7/8ips audio cassettes, the quality of the cassette one bought at the store could be pretty darned close to "sounding" like it was the original master tape -- there was loss all the way through the process, but very little loss that would have translated to the cassette - the cassette was never capable of hearing what may have been on the astoundingly high-quality 30ips tape anyway...

    Thus, your statement is no surprise here.

    In any event, if this is true, one would always try to "bump up" the master "speed" -sample rate/bit rate -at least a step or two ABOVE the "final" rate, in an attempt to avoid loss of quality by "final" format. Thus, a master recording with the ultimate goal of CD, recorded at 24/48 should be good, though at 24/96(Or above) it could(Depending on number of "copies" from master to final) be better.

  3. Berry

    Berry Guest

    That's what happens when you try to shorten a dissertation... Well, basically I do agree with you. I forgot to mention that there was a BIG difference in quality between 16/44.1/48 and 24/48/96, but not a significant difference between 24/48 and 24/96! Even those two guys mention that it COULD(!) be better to choose the highest sample rate available when having a long chain. (BTW., a long chain is never any good). But they sort of WARN that a very high rate could even generate noise and could not be capable of quickly and adequately turning inputs and outputs on and off accordingly to the incoming analog signal. :D
  4. Dave62

    Dave62 Guest

    Doesn't the high end get all messed up at the lower sample rates??
    For instance, a 15khz waveform at 44.1 K will only take 3 samples to map the wavelength( upper and lower limits) and that will be 6 samples at 88.2. In those three samples at 44.1 the wave starts at zero ,peaks positive,hits zero,peaks negative and returns to zero. Even if the waveform were somehow starting exactly on a sample point it would be impossible to map correctly in three sample points because the middle zero crossing would fall exactly half way between two sample points. 96 is twice as good at the higher frequencies but still is far from perfect. If you record tone generated by Protools into protools (eliminating any converter issues) at 15khz and zoom into the waveform it looks a lot more sine waveish at 88.1/96 compared to 44.1/48. Doesn't this make 192 khz ( 12 sample points at 15 khz) far more accurate at the high frequencies? Or am I missing something??
  5. Randyman...

    Randyman... Well-Known Member

    Jun 1, 2003
    Houston, TX
    What you see in your Sample Editor's window IS NOT what the signal looks like when coming out of the D/A. Even IF a waveform starts 1/3 way inbetween a sample point, the waveform will STILL be replicated ACCURATELY on the D/A output. If you analyze the OPUTPUT of a good DA, you'd likely see that there is no difference in a 15KHz sine at 44.1K fs, and a 15KHz sine at 192KHz fs (No, I have not done this myself, but I have read and seen quite a bit of duccumentation on this). Both will be reflected as they were captured (even if the waveform starts IN BETWEEN the sample points). Both will be a smooth sine wave at either sample rate (on the ANALOG OUTPUT). 44.1K will capture anything below 22.05KHz with every bit of accuracy that is possible - including phase.

    This digital stuff is truly mind-boggling. I'm still wet behind the ears, but there are TONS of killer info Here, at PSW, and on the web in general. Nika Aldrich ad George Massenburg (sp) are 2 who surely know their stuff!

    The math involved is just plain crazy IMO. Nyquist Theorem is the final word in Digital IMO (I'm surely NO expert). How it is implemented, and how the analog circuitry interfaces into the digital realm is where the problems occur. High sample rates are NOT the fix IMO.

    Just think - Have you EVER heard a 44.1K CD that sounded good? I have heard quite a few. When it is done right - 44.1K is a beautiful thing.

  6. JoeH

    JoeH Well-Known Member

    Jun 22, 2004
    Philadelphia, PA/ Greenville, DE
    Home Page:
    aggghhhhhhh, not this again........

    It's really pretty simple now; the industry standard has all but settled into a couple of very simple choices:

    1. If $$$ is no object (Big budget movie soundtracks/scores, Audiophile recordings, PTs HD, etc) Use 24bit /192k sample rate. All comparative arguments are moot at that point.

    2. For everything else, use 24/96 if your system (hardware) supports it. Edit and process at this rate, don't go down to the lower rates until your final mix. Then SRC and dither down with the best converters & algorithms possible when making a 16/44 CD, or 16/48 Video Soundtrack.


    3. If your project is going to end up as a video track, start at 24/48 (or 24/96) and them go to 16/48 when finishing up the final video track.

    4. If your project is going to end up as a CD Audio track, start at 24/44 and stay there till the end (you'll lose any subjective "better sound" in the gearboxing process from 48 to 44.) Or again, start at 24/96 and stay there until your mix/bounce/dither to 16/44 at the very end of the project.

    With the speed and power of most DAWs these days, you can start at 24/96 and stay there until you make your DVD, DVD-A or CD final renders.
  7. Dave62

    Dave62 Guest

    Yes it does interpolate and smooth waveforms , but in a complex music waveform, just how accurate can that process be? Its still mathematically filling in the holes. Interpolation and smoothing aside, a high frequency complex wave has to be more accurate at the higher sample frequency because it has had twice or four times as many actual sample points as 44.1. Also, the single biggest stated reason for going to 96 or 192 is that the high end does not sound as brittle. If what Randyman says is true, there should be absolutely no audible difference in highend smoothness between 44.1 and 192, yet IMO there is.

    Can anyone in this forum actually check the output distortion (Thd) of a 15khz sine wave at 44.1 versus 88.2 or 176.4 ?? I am very curious about this and I don't have the tools to check.
  8. TeddyG

    TeddyG Well-Known Member

    Jan 20, 2005
    Dave62 - All of this information/testing/comparison, one would think, someone has done, somewhre - maybe on the web? Maybe not with WL, specifically..?


    Yes, starting high(Initial recording) and ending low(The final format) should be good, but, also yes, the best thing is to do as few "saves", "copys", etc, as possible from first record to finished product. Pre-planning so one does as much of the editing/processing/ as possible with, say, the first "copy", would be great - then to make your next copy "the(Lower) right speed" and the "final product" would be excellent.

    What rates are proper is fun to talk about, where it all ends is a good topic, but it's only going to get more confusing as the "final product" rates get higher! Indeed, like it or not, many of us should at least be looking at 24/192 mastering, just to stay ahead of the curve! Ick! At least hard drives are getting cheaper...

    Whether we can hear the difference between 44.1 and 192, is nearly irrelevent. One doesn't often, if ever "use" the 200+ mile per hour speed of a fancy sports car, but they get bought just the same.

    We must keep in mind though, with our project planning, that it isn't the original we need to plan for(We'll just do that at the highest possible rate we can do well), it's the "final rate", the final format needed that will determine how high we have to start.

    24/48 might be great to start with, for a CD or radio commercial, which will ultimately be 16/44.1 .wav(Or even .mp3, as mine are), but if the final format is 24/96..! it's not great enough... Where are my equipment catalogs!

  9. Nika

    Nika Guest

  10. Nika

    Nika Guest

  11. Nika

    Nika Guest

  12. Nika

    Nika Guest

  13. Nika

    Nika Guest

  14. Costy

    Costy Guest


    What is your point (if there's any)? Have any useful references ?

  15. Nika

    Nika Guest

    What kind of references would you like? I'm not sure what you're looking for, but I'd be happy to help.

  16. Costy

    Costy Guest

    Something in English - a research paper, a doctoral thesis
    (masters may do as well) on the posted subject.


    P.S. Your example with the circle and three points is not complete
    in a sense. You should give three points AND the errors for each
    coorinate. Then we can talk how exact is the circle.
  17. Arrowfan

    Arrowfan Guest

    48khz vs 96 khz

    it all depends on the project. The increased resolution will give more accurate high frequencies and should give tighter bass response.

    Whenever a project involves vocals or drums - anytime I use my C414 mics - I'll record at 96khz.

    I claim to be able to hear the difference. Especially with a vocal group a-capella.

    Bottom line is: if you can afford the extra disk space and CPU overhead that 96khz requires, then by all means use 96khz.

    Th difference is subtle but fine. It can't hurt.

    Most software runs at 32 bit. So its already stepping up your 24 bit audio to 32bit. Why not at least give it 96khz instead of 48khz to chew on as well?
  18. Nika

    Nika Guest

    I'm afraid I do not have a master's or doctoral thesis, though I do have a 400 page primer on digital audio available if that might help? The best place to look for validation on Nyquists theorem would probably be:

    Shannon, Claude E. "Communication in the Presence of Noise." Proceedings of the IRE Vol. 37 (January 1949): 10-21.

    You could also look at the original proof offered by Nyquist at:

    Nyquist, Harry. "Certain Topics in Telegraph Transmission Theory." Transactions of the AIEE Vol 47 (April 1928): 617-644.

    Each of these are fairly short reads and are (in my opinion) a lot easier to get through than doctoral dissertations on the subject. Most dissertations I've read that relate to the audio industry take Nyquist as an assumption, so it is difficult to find something that rehashes the eloquent 11 page proof by Shannon.

    As far as the circle analogy and where you find a flaw, I do not understand. The Nyquist frequency merely tells us what information is required in order to reconstruct the shape. The accuracy with which we acquire it will certainly change the results, and thus the quality of the conversion process is essential. Error in the audio industry as far as A/D conversion is concerned comes primarily from two places: the "vertical" error of bit depth quantization, which manifests itself as low level noise and won't be affected by sample rate changes, and the "horizontal" error of sample timing inaccuracies (jitter) which won't be improved by sampling at a faster rate.

    Please explain more about the breakdown in the correlation between this mathematical analogy. Each explains merely the minimum amount of data necessary with which to represent a particular shape, no?

  19. Dave62

    Dave62 Guest

    Nika, if you weren't so condescending I may have considered buying your book. Not a chance now. I was looking for some answers, not for attitude.
  20. Nika

    Nika Guest


    Yes. I realize. I'm having a really, really awful day. Sorry to take it out here.

    My apologies. When I feel better I'll try to go back and revise my posts to be more helpful.


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