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In Reply to: Upsampling shifts energy to higher frequency ? posted by Busybusy on February 21, 2001 at 20:19:28:
HelloIf you want some nitty gritty information on the subject, may I suggest the following paper:
http://www.mlssa.com/pdf/Upsampling-theory.pdf
Be warned though, it's not a lazy Sunday afternoon read :)
Mart, I really like your ideas...
-Peter
Follow Ups:
Offers at least a plausible explanation though it also proves
that the newfangled upsamplers can always be reduced
to either a standard oversampling filter or a Wadia-like
interpolation filter. I.e. old stuff.
TNT Audio Netzine
I'm not sure about describing upsampling as a slow roll off digital filter by default. The tests on the CS8420 show that it produces significant ringing, indicating that the cut off is pretty sharp. It's stopband attenuation is around 110dB.Also, saying that the time smear introduced by digital filtering is inaudible because speakers are worse in this respect doesn't really hold up. Speakers also produce "large" quantities of harmonic distortion, yet lesser distortion in eg. an amplifier is surely audible too.
I don't want to sound too negative though - ultrasonic noise/signal could well have a psycho-acoustic effect.
As a general thought, many CD players with high levels of noise and distrtion are often reviewed as having more "life" and realism.
Daniel Espley
Oh, certainly not by default indeed. Prior to reading that
article I had assumed that all upsamplers would be high-order
filters, i.e. conceptually identical to oversampling filters.The paper opens the possibility that at least some upsamplers
have a slower roll-off, and the following track about this
helping spread out DNL is an interesting one.But it still doesn't offer a plausible mechanism for delta-sigma
DACs. And neither does it for dCS's RingDAC, which is by itself
a cluster of statistically-averaging current sources and hence
should need no other tricks to reduce its DNL below thermal
noise effects.
TNT Audio Netzine
Hi Werner,The CS8420 almost certainly uses a brick wall filter, but I believe the DCS gear allows various co-efficients. But, as you said previously, we are now dealing with a different issue than upsampling's validity on it's own.
Dan
But the only purpose of upsampling (a 16/44.1 PCM CD) is to allow the use of an alternative slope filter rather than a steep brickwall filter.
Hi Garth,the purpose of sample rate conversion (upsampling) is to make the output data asynchronos with the input data. Windowing the impulse with various coefficients is a technique that is not unique to upsampling - it is regularly doen with oversampling.
dan
> > the purpose of sample rate conversion (upsampling) is to make the output data asynchronos with the input data < <Perpetual Technologies website mentions an 88.2 upsampler for the pro market. I wonder how sales are going. :~[
> > the purpose of sample rate conversion (upsampling) is to make the output data asynchronos with the input data. < <IMO the purpose of upsampling is to market to audiophiles who make buying decisions based on bigger numbers (24 is bigger than 20, 96 is bigger than 44.1) ;~)
Dan Bonhomme
Hi 'Fan,There's nothing wrong with upsampling - it has a legitimate use, and is an effective jitter reducer depending on how well it is implemented. I think the trouble perhaps started when someone supposed that, for instance, upsamling to 96KHz improved transient response to the equivalent of true 96KHz data. DCS have done nothing to discourage this myth, mind. Why would they?
Dan
> > There's nothing wrong with upsampling < <I believe there is something wrong with upsampling. It's too complicated. Any benefit achievable through upsampling in a DAC can be more efficiently and effectively implemented with I2S connection, oversampling and dither. And these avoid the questionable algorithms required for non-integer conversion.
If upsampling were done in multiples of 44.1, I'd be far less skeptical of their reasoning. Why convert to 96 instead of 88.2 kHz, or 192 instead of 176.4 kHz? The only reason that makes sense to me is that they are trying to capitalize on the popularity of the true higher sampling rate mediums, DVD and DVD-A.
Dan Bonhomme
Hi Audio Fan,SRC was designed to provide compatibility between digital devices that run at different sample rates. It does this flawlessly (the two linked devices are now 100% compatible), which is what I meant when I said "There's nothing wrong with upsampling" - as a concept, it does the job it was designed for.
My understanding is that as the initial sample rate will be modified by jitter, as pointed out by Steve below, there is no such thing as true integer conversion. However, the more samples it uses to define its "from" sample rate, the more closely it can calculate the ideal data value.
Daniel Espley
That's OK. What say we start the thread again up top just to bug everyone? ;o)Dan
Posted by Daniel Espley on February 27, 2001 at 13:55:14:
In Reply to: Re: a decent paper indeed posted by Audio Fan on February 26, 2001 at 14:03:11:
Hi 'Fan,
There's nothing wrong with upsampling - it has a legitimate use, and is an effective jitter reducer depending on how well it is implemented. I think the trouble perhaps started when someone supposed that, for instance, upsamling to 96KHz improved transient response to the equivalent of true 96KHz data. DCS have done nothing to discourage this myth, mind. Why would they?Dan
Partially true but dCS has consistently stated that upsampled data is not "true" 96 or 192 sampled data and that in fact there is no difference in the frequency response comparing the original to the upsampled. The differences are in the filters used and they provide a choice of six.
Hi Garth,I read an article in HiFi world when the DCS SRC first came out. The suggestion put to DCS was that upsampling improved transient response, and they were happy to play along. There have been a few similar articles, including one quite recently that someone here supplied a link to (I can't find it). None of this means that DCS don't make excellent sounding equipment, mind.
Daniel Espley
Hi Daniel!Like most thngs in audio one should simply evaluate the product for it's sonic results. Even dCS admits they don't know all the answers as to why upsampling (as they impliment it) has the sonic effects it does at least with their DACs. I have tried the Purcell with other 24/96 capable DACs and the difference to 16/44.1 was not that significant. However with the dCS DAC the difference is quite noticable as are the different filters which are not available at 16/44.1.
> > Like most thngs in audio one should simply evaluate the product for it's sonic results. < <I agree. Unfortunately I see a distinct trend on this board to substitute silly numbers games for system matching and personal listening experiences. Some people are buying digital gear simply because it upsamples to higher numbers, which they assume is better. And hype on this board facilitates the nonsense.
Eliminating CD players from one's short list just because they don't have a 24/96 DAC or upsampling is ludicrous. Buying gear by the numbers without critical evaluation and comparisons in one's own system is not the audiophile way. Yet is seems to be common, especially with net commerce.
Dan Bonhomme
Daniel wrote:> There's nothing wrong with upsampling - it has a legitimate use, and is an effective jitter reducer depending on how well it is implemented.
I gotta disagree a little with this point, Daniel.
Normally, jitter may be considered as the distortion in the time domain vs. the amplitude of the signal being reconstructed. Recovered clocks mess with this timing, so that, if you clean up the clock jitter, then the correct amplitude can be reconstructed at the correct time.
In an ASRC type of polyphase upsampler (AD1890, CS8420), the input data is *rewritten* using the output sampling frequency as its masterclock. In otherwords, all the jitter inherent in the input is now encoded (and lost as measureable "jitter") in the serial data.This can provide a very clean clock timing to the following digital filter or DAC, but the input to the DF or DAC (even if downsampled) is not bit for bit the same as the original. You have traded an increase in precision in the time domain for a distortion of amplitude being reconstructed.
The measurable clock (that times the serial data) "jitter" is indeed lower, but the recovered input clock jitter is now contained (and lost)in the serial data.The correct timing of the wrong amplitude is the result.
Cheers,
-Steve
> > The correct timing of the wrong amplitude is the result. < <Any idea of how that would sound? Would it be less objectionable than jitter?
Dan Bonhomme
I haven't compared and haven't investigated further.I've gone with 1x Oversampling.
Cheers,
-Steve
Hi Steve,you wrote:
"The measurable clock (that times the serial data) "jitter" is indeed lower, but the recovered input clock jitter is now contained (and lost)in the serial data."Well, we're not really disagreeing at all - the jitter is lower.
I understand that with the ASRC process jitter is encoded into the data signal. The degree to which this happens is dependant on the quality of the implementation. For instance, working with 24 bits will help.
So, to put is crudely, one type of type of distortion is traded for another. The effectiveness of using ASRC can therefore be determined by whether it improves dynamic range over an alternative method of digital interfacing.
Daniel Espley
"Longer, lower, wider..." in the digital marketplace.Cheers,
-Steve
As a general thought, many CD players with high levels of noise and distrtion are often reviewed as having more "life" and realism.And some have conjectured that the high levels of ultrasonic noise inherent in SACD playback adds some of that extra realism as well which may have some credence since adding in some extra noise filtering in the digital domain via the "normal" switch on the first batch of players without significantly changing the frequency response causes a very real degradation in perceived sound quality.
Just changing the filter slope on a digital filter such as the DF1704 (or to a lesser extent with the CD players that have multiple filter coefficients) and allowing alias products and associated noise components through very much opens the sound on CD playback. Many factors are at work that are not easily explained.
Dave
Hi Dave,isn't it a slightly ironic propostition that our ears want to hear some 20KHz plus information, even if the information is rubbish? I guess I find it a little amusing as the general direction of improving fidelity is to widen dynamic range - but now we are wondering if the opposite is the case (in this respect).
I would think that genuine ultrasonic information, as supplied by hi-res formats would be supperior still, but who knows? It also bolsters the case for super-tweeters, even when using a non-oversampling DAC.
Dan
Daniel wrote:> I would think that genuine ultrasonic information, as supplied by hi-res formats would be supperior still, but who knows?
Since we are dealing (mostly) with harmonic overtones (and not true fundamentals) in the ultrasonic region, maybe the discrete reproduction of these is not (perceptibly) important. Maybe a highly corelated noise component (alias distortiion) will add a sense of "naturalness" that a BW limited signal (to kill this noise component) will not.
> It also bolsters the case for super-tweeters, even when using a non-oversampling DAC.
Could be.
But with using a 44.1kHz sampling frequency, in the upper octave of reconstruction (1/4-1/2Fs or 11.025kHz-22.05kHz), the only thing that can be "reassembled" is a pure sine wave.
Maybe by "dirtying" up this with a correlated noise will impart an assymmetry to the waveform, that is more closly percieved as "natural" without the necessity of a wider bandwith.
Just speculatin'....
Cheers,
-Steve
Hi Steve,I appreciate what you are saying, but I suspect most violin makers for instance would feel that upper harmonics are rather important. Also, reasearch suggests that our ears locate sounds by the leading edge (transient), and are sensetive well beyond 20KHz. I think all in all, most would prefer true ultrasonic information if given the choice.
> Just speculatin'
Me too!
Dan
I've wondered if the ear wants to hear the products of the interaction of ultrasonic frequencies with sonic frequencies, rather than the ultrasonics themselves (which by definition are out of hearing range).Dan Bonhomme
Dave wrote:> Just changing the filter slope on a digital filter such as the DF1704 (or to a lesser extent with the CD players that have multiple filter coefficients) and allowing alias products and associated noise components through very much opens the sound on CD playback.
Or better yet IMO, getting rid of the digital filter altogether as in the Audio Note and Lab 47 non-oversampling DACs. The aliasing *problems* (more in theory than in the listening) are still there, but it does also eliminate the time domain distortion (pre and post echos -Gibbs Phenomena) that all FIR digital filters add.
> Many factors are at work that are not easily explained.
Exactly. :-)
Cheers,
-Steve
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