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In Reply to: RE: Chord up-samples? posted by knewton on June 20, 2017 at 17:49:52
When performing synchronous interpolation ("upsampling" or "oversampling"), generally the original data points are still present (although often scaled in value), and additional interpolated points are calculated between the already-existing data points.
In contrast when performing asynchronous interpolation, virtually *all* of the original data points are discarded and replaced with calculated values. In the very best case of a common asynchronous rate of 44,100 to 48,000 samples per second, there would only be one original data point every 147 or 160 samples (depending on the direction one is converting), as those are the least common multiples of the two rates. (In practice this has never been achieved as doing so would require interpolating by either 147 or 160 and subsequently dividing by 160 or 147. The resultant intermediate frequency of 7.056MHz is too great and capable chips priced too high to be available in any known equipment. It is past double-DSD rates but would require 64-bit accumulators rather than the single bit used in DSD.)
Instead what is used is an ASRC (often also used as a "jitter eliminator") wherein the incoming and outgoing rates have no correlation whatsoever. In that case, every single output sample is an *interpolated* value. While each output sample is *related* to the input samples, the relationship is known only to the designer of the particular algorithm used by the ASRC chip.
My personal experience, and what appears to be a broad general consensus, is that synchronous interpolation (wherein the original sample points are retained or scaled) is sonically preferable to asynchronous interpolation (wherein the original sample points are completely replaced by interpolated values).
As always, solely my personal opinions, prone to error, and not necessarily those of my employer or pool-boy.
Maybe the Chord DAC upsamples to the lowest common multiple of the popular audiophile sampling frequencies like the PS Audio DirectStream, e.g. 28.224 MHz.
Here's a question from a curious enthusiast: I thought with interpolation by an integer factor, the original data points will still get modified a bit since there's no such thing as a perfect low pass filter. However, you can use half-band filters to 'retain the original samples', which seems to be the approach used by Schiit. What are the tradeoffs involved between the former and the latter?
Charles, as you correctly surmised, I was addressing synchronous interpolation digital filtering in my upthread comments. I don't disagree with your pointed criticism of ASRC. As I understand it, the main problem with ASRC solutions is the operation of the 'ratio estimator' block. This block is tasked with determining the ratio of the input to output sample rate. The resulting ratio estimation is computed by taking a running average of the two rates, which can vary or drift slightly in value over time and thus provoke artifacts. In addition, very small rate differences, where the input and output rates are nearly, but not quite, the same can also provoke ratio estimator artifacts.
Artifacts appear because the programmable filter coefficients (which are derived from the ratio estimator's computations) utilized in the rate conversion/interpolation polyphase FIR filter block are themselves interpolated values. This is done in order to compute coefficients which support arbitrary input/output ratios. One of the problem aspects of ASRC design seems to be insufficient polyphase filter coefficient precision, which is magnified in hardware based IC converters due to limited hardware resources.
Just my very much non-authoritative understanding of the underlying problems. Perhaps, someone with ASRC design expertise will add to, or correct my above basic assessment.
"Instead what is used is an ASRC (often also used as a 'jitter eliminator')"
ASRC does not eliminate jitter, the input jitter gets encoded as "amplitude" errors in the converted signal. (The converted signal might not necessarily have less jitter.) If the same signal were to be converted in a later trial, a totally different set of "amplitude errors" would be encoded in the converted signal. The converted data in multiple trials would never be numerically identical.
In contrast, synchronous conversion does not introduce "amplitude errors", because timing is not involved in the conversion. Just the values of each input sample. The converted signal in multiple trials would be numerically identical (provided no data is misread from the media).
"My personal experience, and what appears to be a broad general consensus, is that synchronous interpolation (wherein the original sample points are retained or scaled) is sonically preferable to asynchronous interpolation (wherein the original sample points are completely replaced by interpolated values)."
This is because with ASRC, the amplitude errors from the input jitter embedded in the converted signal is essentially noise introduced into the signal. Synchronous conversion does not have this problem.
> > ASRC does not eliminate jitter, the input jitter gets encoded as "amplitude" errors in the converted signal. < <
I must admit to never having studied the underlying theory of ASRC's in great depth, simply because I've not liked the sound of any I've tried. Knewton's post just above (in Classic view) has a fair amount of technical information that seems accurate to me.
All I can say is that whatever the (real, perceived, or imagined) shortcomings of ASRC's may be, that they work well enough to fool present day measurement technology. I have performed careful comparisons with and without the ASRC, and an ASRC can easily make any DAC implementation (whether poorly or perfectly executed) and yield essentially perfect results on the JTest as used by both Stereophile and Hi-Fi News (and I believe also at least one German publication).
In my opinion ASRC's are the chief reason that one sees virtually "textbook perfect" result in many JTest measurements for the past few years, even on sub-$300 DACs. It also means that listening is even more important than ever. Before ASRC's became widely available, there used to be at least a weak correlation between the JTest measurement (which could show the quality of implementation of a DAC) and sound quality. Currently there can be an inverse correlation (in my personal opinion), as I've yet to hear an ASRC that sounds musically natural, yet they all measure essentially perfectly with the JTest.
As always, strictly my personal opinion and not necessarily that of my employer or lap dog.
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