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In Reply to: RE: Just a thought ... posted by Jim Austin on June 11, 2017 at 13:55:59
"time smear" goes back to the first tape recorders. Wow and flutter. Nothing new.
Many of today's DACs have virtually unmeasurable jitter. There are numerous non proprietary ways to deal with it, including synchronous upsampling like Bryston does.
> > Many of today's DACs have virtually unmeasurable jitter. There are numerous non proprietary ways to deal with it, including synchronous upsampling < <
1) There are only two magazines (both print) of which I am aware that publish jitter tests - Stereophile in the US and Hi-Fi News in the UK.
2) Both have upgraded their measurement equipment at least once, making it impossible to compare earlier tests from their more recent tests. Specifically Stereophile was the first to measure jitter, but using a unit from Ed Meitner. The difficulty with this unit was that it required opening up the unit and connecting test leads to specific pins on the DAC chip itself. This wasn't so bad in the early '90s as all of the parts were large through-hole devices and most of them were R-2R "ladder" DACs that were sensitive to jitter on the word-clock pin.
By the end of the '90s there were new DACs, almost all surface-mount devices that are extremely difficult to probe without damaging the DUT. In addition the ladder DAC chips were changed such that jitter was only important on the bit-clock pin, and new delta-sigma DAC chips were available that are sensitive to jitter on the master-clock input pin (that drives the modulators in the output stage). All of this led Stereophile to switch to a new analyzer, developed by Paul Miller (currently editor of Hi-Fi News). This machine obviated the need to open up the DUT, as it only measures the analog output signals. It is made from plug-in A/D cards made by National Istruments that fit into a desktop computer, along with audio specific test routines developed by Miller. The problem with that early machine was the A/D converters used were only 16-bits and they were inside an extremely noisy environment (a desktop computer with dozens of high-speed clocks and switching power supplies). Jitter tests made with this machine typically show artifacts of the test equipment in addition to actual jitter in the DUT.
Many years ago Stereophile switched to an Audio Precision which has a noise floor low enough to accurately measure audio equipment without the artifacts of the test equipment interfering. Hi-Fi News still uses Paul Miller's equipment, but at some point the National Instruments cards were upgraded to use 24-bit A/D converters and improved shielding. It's unclear if that setup performs equally to Stereophile's Audio Precision, but it likely is quite close.
3) In addition to the constantly improving measurement equipment, there has been a parallel trend in many of the D/A converters that are tested. Specifically they often incorporate Asynchronous Sample Rate Converters (ASRC), either separately or built into many modern DAC chips. When this ASRC is employed, almost any D/A converter will exhibit "textbook perfect" results on the J-TEST jitter test used by both Stereophile and Hi-Fi News. However my personal experience is that ASRC can dramatically improve the measured performance of a converter while at the same time significantly degrading its audible performance. YMMV.
4) Unlike ASRC, so-called "synchronous upsampling" does nothing to reduce jitter (measured or audible). It can affect the sound quality, as it is simply a specific type of digital filter. As such they will all affect the sound quality differently depending on the parameters of the filters used.
The bottom line is that I agree that when comparing jitter measurement from now to 10 (or even 20) years ago that it definitely appears that the equipment has improved. I am much less clear that most equipment has actually improved when it comes to audible jitter, especially when ASRC is employed.
As always, strictly my personal opinion and not necessarily that of my employer or pharmacist.
Very educational thank you.
I stand corrected on synchronous upsampling. Interesting, the Bryston DACs, engaging it makes the sound a bit softer, but very pleasing, and a bit more textured. This is across the board.
The upsampling is user select-able. I think that is a nice idea.
> > Interesting, the Bryston DACs, engaging it makes the sound a bit softer, but very pleasing, and a bit more textured. < <
It appears that you are referring to the Bryston BDA-3. According to the owner's manual, "the internal sample rate converter upsamples incoming 44.1kHz and 88.2kHz digital audio to 176.4kHz. All 48kHz and 96kHz digital audio upsamples to 192kHz. The Upsample feature does not affect HDMI or USB."
That unit uses dual mono AKM AK4490 DAC chips, which have built-in 8x digital "oversampling" filters. (The correct technical term is interpolation filters.) When "upsampling" is engaged (again, the correct technical term is "interpolation"), a separate interpolating digital filter is inserted prior to the interpolating digital filter built into the DAC chip - nothing more and nothing less. There are three things to note in this situation:
1) Concatenating digital filters is done all the time, almost always to save money. Virtually all 8x interpolating digital filters are a concatenation of three 2x interpolating digital filters. Nothing new here.
2) The first digital filter in the chain has the greatest sonic impact. It is possible that the sonic differences are simply due to the different characteristics between the first stages of the external ("upsampling") and internal ("oversampling") interpolating digital filters.
3) The other factor that may come into play is the rate at which the modulators in the DAC chip are operating. Depending on the specific internal architecture of the DAC chip, the modulators may or may not be operating at different rates when presented with different input rates (eg, single-, dual-, or quad-rate signals). I've not seen any that are so affected (the modulators typically operate at the master clock rate set by the local crystal oscillator), but it is conceivable that there are exceptions with which I am unfamiliar.
The bottom line is that the "upsampling" feature inserts an extra digital filter into the signal chain. This is very much the situation with MQA, as well. The degree to which the sound is affected (for better or worse) in either situation is simply due to the effect of the extra digital filter.
Just out of curiosity, how would you describe the magnitude of the sonic difference created by the addition of the "upsampling" digital filter in the Bryston DAC?
As always, strictly my own personal opinions and not necessarily those of my employer or other digital engineers.
Ok, I have no problem admitting that some of the dac signal processing you lay out here is over my head!
But to answer your question, on the BDA-3 DAC, the difference in sound when the upsampling is engaged is pretty stark. It is almost like the signal is being passed through a lush tube buffer. Better? Definitely different.
I also wonder how hardware upsampling like the Bryston scheme differs from upsampling at the server stage, like with Roon, or even HQplayer. Upsampling to DSD was in HQPlayer was a big fad recently.
Now with Roon providing that capability along with the previously discussed powerful DSP tools they provided with the last update, to me it looks like MQA is more obsolete with every week that goes by.
EDIT: I am also reminded of the SONY HAP players, which have a user engagable "Remastering" process. Digital filtering of course.
> > Upsampling to DSD was in HQPlayer was a big fad recently. < <
There is a potential for this to improve the performance of some particular implementations of delta-sigma DAC chips. Depending on the internal architecture, it is possible to run the modulator on the output stage at higher rates, which will definitely change the sound - possibly for the better.
As always, strictly my personal opinion and not necessarily those of my employer or baby-sitter.
My master 7 dac can be run nos up to 8 times oversampling. I prefer nos as it sounds more natural than oversampling
> > I prefer nos as it sounds more natural than oversampling < <
I would suggest that it all boils down to the particular oversampling (interpolation) filter used. A decade ago there were many DACs introduced with NOS (filterless) designs. I was curious and when Ayre developed the ability to create custom digital filters, the first test we tried was NOS. This replaced a combination of an external 4x "upsampling" filter feeding the 8x "oversampling" filter built into the DAC chip.
There were significant improvements in many areas - the midrange in particular was very pure and natural, but the frequency extremes seemed to be not quite up to the performance level of the broad midrange band (~200Hz to ~5kHz). I then went down a rabbit hole of various interpolation rates (4x, 8x, and 16x), window functions (Kaiser, Taylor, Gaussian, etc.), multiple parameters affecting the shape of the rolloff curve, and finally various dithering algorithms.
In the end I felt that we had improved upon all of the sonic advantages of NOS without losing anything. But I would agree that in general NOS (filterless) is an overall improvement over the typical filters built into DAC chips. YMMV.
As always, strictly my personal opinions and not necessarily those of my employer or Pee-Wee Herman.
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