I agree with Onhwy61 - article is a nonsense. First, motion that 16/44 is perfect if meets Nyquist criteria is first nonsense. Nyquist criteria applies only to continuous waves. Short high frequency bursts like cymbals will suffer the most of distortion. Second nonsense is that digital filter can suppress 96dB within 4kHz without any problem. There is no filter in the world, digital or analog, that can do that (no matter how many poles) with even group delays (or linear phase if you prefer). Uneven group delays will cause poor summing of harmonics (delayed differently) and change in sound. Reducing suppression won't help since low level signals above 24kHz will "fold" into audible band starting at 0Hz. Next nonsense is that ultrasonic frequency is harmful to the ear and modulate tweeter. Not only that 192kHz is WAY easier to completely filter out than 44kHz but also modulation can only happen on nonlinear element and for this to happen membrane has to move - not likely at 192kHz (even if your amp and CDP have such bandwidth). Then he claims that higher resolution does not increase dynamic range because of ambient noise floor forgetting that it is still improving resolution for louder signals. He claims that oversampling can increase resolution and sampling rate - true, but it is done with interpolated samples while 24/192 contains real samples. I agree that we might have hard time to hear better above certain resolution/rate, for instance 20/96 but claim that 24/192 is harmful is complete nonsense. |
Audiofreak32, I enjoy my system all the time. ALAC is lossless while wireless transmission is bit perfect. Benchmark is as clean as it gets on the verge of being sterile but it fits perfectly with my warm sounding Hyperion HPS-938 speakers. In addition my Benchmark is modified with better sounding op-amps. Sure I could be doing much better but for much more $$$$.
Linn looks very impressive but it is $7500 while Benchmark + AE were $1100 total. I'm perfectly happy with 16/44 limitation of AE, having over 1500 redbook CDs on HD. Computer costs me nothing since I already have one. My setup also requires only one pair of ICs. |
"nonsense! The Nyquist criteria applies to any signal that needs to be quantized. The Nyquist criteria only gives the minimum requirement; it does not say that one is forced to have only 2 samples per highest frequency."
Yes, you can have more samples (for instance 192kHz) but he claims that 44.1kHz (two samples) is all you need.
Again, Nyquist applies to continuous waves ONLY.
from Wikipedia:
"The theorem assumes an idealization of any real-world situation, as it only applies to signals that are sampled for infinite time; any time-limited x(t) cannot be perfectly bandlimited."
Perfect reconstruction of continuous signals close to Nyquist frequency (for instance 15-20kHz) is possible but when signals become very short, reconstruction is much less than perfect.
As for filters - look at typical response of 2and 8 pole 20kHz Bessel filter in dB:
2pole 8pole
20kHz -3 -3
22kHz -3.63 -3.67
40kHz -9.82 -13.68
80kHz -20.32 -51.81
As you can see there is very little attenuation difference at 44.1kHz/2=22kHz with 4x higher number of poles. You would perhaps need hundreds of poles and still not get -96dB. Dramatic difference shows at higher frequencies beyond the "knee" of the filter (160dB vs 40dB per decade). Whole purpose of converting analog to digital at higher rate is to represent bandwidth of 20kHz more accurately and not to extend bandwidth. Downsampling 24/192 master tapes to 16/44 removes some information, (audible or not) but to claim that 24/192 is inferior to 16/44 is complete nonsense.
As for dynamic range, again the point is resolution of the signals above noise floor. According to this article if I listen at 85dB peak and have 35dB ambient noise at home I should not be able to tell the difference between 16 and 8 bit recording (corresponding to about 50dB range). That's nonsense as well.
What about 192kHz being harmful? It doesn't get more silly than that. |
Al, What sounds inconceivable to me is that 24/192 recording supposed to gain sound quality by downsampling it to 16/44 to be upsampled again, perhaps to the same 24/192. Am I reading it right? Is downsampling + upsampling somehow improving sound by replacing real samples with artificial interpolated samples and recreating same harmful 192kHz? |
Al, On one hand he has whole chapter titled "192kHz considered harmful" describing harm that 192kHz can do to amplifiers and speakers to later say this about oversampling:
"This means we can use low rate 44.1kHz or 48kHz audio with all the fidelity benefits of 192kHz or higher sampling (smooth frequency response, low aliasing) and none of the drawbacks (ultrasonics that cause intermodulation distortion, wasted space)."
According to above when DAC is exposed to 192kHz sample rate from CD it is harmful to all analog circuitry afterwards (including power amp and speaker) but when DAC upsamples redbook CD (24/192 downmixed to 16/44), the same 24/192 becomes benign. In either case DAC outputs samples at 192kHz rate but in one case it is less harmful to analog circuitry? |
Al, I wonder if 24/192 contains any ultrasonic frequency at all. Why would they leave it preparing hi-rez files? Where this ultrasonic frequency comes from? Again, notion that 192kHz sampling is harmful is a little farfetched. Do we have any studio sound engineers on our forum that could explain it to us?
Bombaywalla, Thanks for the info on filters. I'm dealing mostly with 4-tap lowpass FIR filters at work but 500-tap filter is really something. One graph shows interesting step response typical to most of CDPs with ringing appearing before and after the pulse. That might affect the sound since our ears are very sensitive to it. Stereophile posted similar test results comparing apodizing and non-apodizing filters. In comparison there is no antialias filters used in SACD creation making better, more natural step response (transients).
I do not have golden ears and like the sound of my system very much but just believe that processing back and forth 24/192=>16/44=>24/192 is not likely to improve anything. Higher sample rates are not to extend bandwidth but rather improve filter response reducing pre-echo effect. Apodizing (windowing) filters, available in few CDPs like Meridian, allow to eliminate pre-cho completely but AFAIK are not suitable for 44.1kHz because there is not enough space between 20kHz passband and first alias to fit filter's windowing function. DSP processing is not my field of expertise but even if everything looks peachy in frequency domain there is a lot to be improved, possibly by higher sampling rate, in time domain (transient response).
Audiofreak32, technical articles are to understand better what is happening but you're right, that at the end what counts is listening experience. At the level of 20/96 or 24/192 placebo (or negative placebo) effect might be a dominating factor. Just the fact that I feel good about my gear can make it sound better to me than to others.
Oldears, Choice or audibility of different formats might depend on setup. In my setup, for instance data is wirelessly delivered ALAC compressed to Airport Express and contains no timing. It is also bit perfect. Lack of timing is important because it eliminates any influence of computer processing or playback program, computer noise, etc. At this point timing is recreated in AE and data is streamed to Benchmark DAC1 with low 258ps jitter further suppressed by Benchmark processing. I could also save data in other formats but it would eat up some processing power of my computer that I use for other chores (like typing this). |
Al, My DAC and perhaps a lot of them cuts at 45kHz. AFAIK vinyl recordings also extend to about 50kHz. It is considered to be an advantage of vinyl. |
Bombaywalla, Making amplifiers cutoff frequency at 20kHz means that phase shift at this frequency will be in order of 45 degree causing bad summing of harmonics. My small Rowland 102 amp has 65kHz bandwidth with about 22deg phase shift at 20kHz. New Rowland amp model 625 ($15k) has bandwidth of 350kHz. Mr. Jeff Rowland knows what he's doing (I'd like to think). As for 50kHz - a lot of power amps have -3dB bandwidth of 50kHz:
All Atmasphere amps: >100kHz
All Rowland Amps: >65kHz
All Cambridge Audio Amps: >50kHz
All Krell Amps >95kHz
All Classe Delta Series: >100kHz
All Classe CT series: >80kHz
All Luxman Amps: >100kHz
All Parasound Halo Amps >100kHz
All Parasound NewClassic Amps >50kHz
etc.
But if you won't to spend 2nd mortgage you'll find amps like:
MBL Reference 9011: 320kHz
Goldmund Mimesis 8: 800kHz +/-1dB
Soulution 710: 1MHz |
Bombaywalla, Bifwynne - No, I don't know the numbers, but suspect it is at about a half or less. Delivering full power at high frequencies is not really important since very little power goes to tweeter. In my Rowland 102 max power at high frequencies would damage amplifier (burn out output choke). Icepower module 200ASC used in my amp is specified at 200W at 10Hz-20kHz but it is only momentary power. FTC rated power is specified as 55W but only up to 8kHz, with warning about damage to the choke at higher frequencies. It is not really important because average power when playing music is only few percent of peak power. The reason for that is that if on average music has half of peak loudness, it means 1/10 of power (logarithmic scale) and then music also has gaps (unless one listens to sinewaves). 55W of power at any frequency above 8kHz would most likely damage any tweeter, not to mention hearing. What worries me a little is 22 deg phase shift at 20kHz (-3dB bandwidth is 65kHz). It would weaken upper harmonics summing. My amp would benefit from a little more "air" but it might also be my hearing (not getting any younger) or the fact that speaker has warm character and is never bright - even on worst CDs. It has, in the system, very clean, pronounced natural sibilants. I don't want to change it and before I audition another amp (like Rowland 625) I need to fix room acoustics.
I understand Rowland's idea behind 350kHz bandwidth in model 625 - no phase shift at 20kHz and perfect step response but 1MHz bandwidth of Soulution 710 is perhaps too much. According to reviews it is excellent amplifier with very little negative feedback but in general the easiest method to improve most of amplifier's spects (like THD, IMD, DF, Bandwidth) is to use deep negative feedback that also enhances odd harmonics (overshoot in time domain caused by amps signal delay and thus late feedback summing - known as TIM) making unpleasant bright sound that SS amps are famous for. Class D has small advantage here, being practically one stage (little delay). Less than perfect design of such 1MHz amp can cause problems including instability followed by oscillations that can damage speakers or sensitivity to RFI. I would tend to agree with Audiofreek32 that numbers are not that important and often amplifier with the best spects has the worst sound. Selecting gear for audition by company reputation or designer's name makes more sense to me. |
Thank you Al. If there are any harmonics within 50kHz amplifiers should be still linear (modulation can only happen on nonlinear element). Any problem would already show with vinyl gear that has similar bandwidth. I don't really see any source of audible harmonics above it. On one hand studio engineers would clean it up but on the other microphones already do it. Most of microphones go only to 30kHz and some extended response go to 50kHz. One of the most popular Neumann U87 ($3600) goes only to 20kHz while the most expensive I could find Sony C-800G ($8000) is only 18kHz. At the concerts as well as in the studio nothing goes directly - everything comes thru microphones to PA system or studio console. Further more, I suspect that studio equipment bandwidth does not extend any higher providing natural filters as well. |
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