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In Reply to: RE: Thanks... posted by Charles Hansen on May 23, 2017 at 16:28:12
Thanks Charles--interesting. I'll add only that, as you know since you've clearly read the patents--some/much of what MQA does is done on the transmission side; coding and decoding work together. My impression from reading the patent--I still lack the expertise to read it well--is that this offers advantages over controlling only the receiving side; is there perhaps an analogy to vinyl/RIAA here?
Posted by Charles Hansen (M) on May 28, 2017 at 12:17:48
> > is there perhaps an analogy to vinyl/RIAA here? < <
I don't see how there could be. Pre-emphasis and subsequent de-emphasis is virtually mandatory for phonograph because of the underlying physics of the transducers. The Redbook CD specification also allows for pre-emphasis/de-emphasis, and it provides a slight advantage (about 1.5 bits extra resolution in the top octave) due to the typical spectral content of music. It seems that it's more trouble than it's worth, as only a handful of very early CDs employed pre-emphasis.
MQA is claiming to do something completely different - specifically, "correct" for "timing errors" created during the original A/D conversion process (so-called "de-blurring"). These "timing errors" are actually artifacts of the steep anti-aliasing filters with sharp "knees" at the corner frequency. The only thing that can be done to "correct" these "timing errors" ("de-blur") is to filter out the "ringing" created by the filter and *hope* that the new filter doesn't create worse artifacts. There are *many, many* digital filters that reduce the artifacts created on the A/D side, the first being from Wadia in the late 1980s. MQA is far from breaking any new ground in this area.
A fundamentally better approach would be to use an A/D converter that does *not* create artifacts during conversion. All DSD A/D converters are free from this problem, as they do not require any anti-aliasing filters at all. However DSD creates a new problem in that it is impossible to process the signal (change volume levels, mix, EQ, and so forth) without first converting to PCM. Conversion to PCM is done with anti-aliasing filters, so the problem springs back to life (think Whack-a-Mole here). Plus each conversion back to DSD adds additional noise.
I believe the best solution is to use true high-res PCM (trivially easy to post-process), but use digital filters on both ends (anti-aliasing for A/D and reconstruction for D/A) that don't introduce sonic degradation. The Ayre QA-9 ADC does exactly that (for dual- and quad-rates only - the single rate minimizes the artifacts but cannot eliminate them). Many companies have created DACs with special filters designed to minimize playback artifacts and also reduce the artifacts from the A/D converter (beginning with Wadia in the late 1980s). There are many companies that have followed the path that Wadia created, or in a few cases pushed that envelope even further.
As always, my postings reflect only my personal opinions and not necessarily those of my employer or son's pet snake.
Charles, thanks very much for this. MQA claims an end-to-end (which I take to mean encompassing the ADC and the DAC) "impulse response" with a main lobe of duration ~ 6 samples, with "ringing" limited to just one negative-going overshoot (ok, undershoot) "lobe" with no more than 10% of the total area (is that with linear or log scales? I don't remember). As far as I know they've released no evidence that they're actually achieving this (I asked for it once more than a year ago; go the typical audiophile "trust your ears" response, which I think I've heard elsewhere ;-) ), so feel free to comment on this aspect of things. But the main question:
When you say the QA/B-9 combo does NO damage at dual or quad (I assume that's 88/2/96 or 176.4/192?), what do you mean exactly? To adopt the MQA vocabulary, I would take to mean that an impulse equivalent to one sample wide (say, 5 microseconds at 192) would, after ADC/DAC, still have a width of 5 microseconds--is that what you're saying the QA/B chain can achieve? (I realize I so far haven't mentioned the question of the POSITION of the impulse in time--phase if you will--which could have uncertainty even if the width is preserved.)
I'm simply trying to understand what you mean when you say the QA/B chain does no damage, and how to relate that to MQA's claims.
I hope this line of inquiry makes sense.
When an impulse passes through any band-limited system, analog or digital (which therefore includes every non-imaginary system), the impulse will be necessarily spread over time. MQA's marketing material shows how air acts as a low-pass filter - the farther the signal travels through the air, the more it is stretched in time. This is equivalent to saying "the farther the signal travels through air, the more the high frequencies are attenuated", yes?
Conversely a wider bandwidth system can pass an impulse with less spreading of impulses. With digital systems, the upper bandwidth is set by the sampling rate. With analog systems the upper limit is the concatenation of the responses of all of the stages in the chain, starting with the recording microphone and ending with the playback loudspeaker. In general analog systems have a wider bandwidth that does single-rate digital - otherwise there would be no need for an anti-aliasing filter in an A/D converter.
The Ayre QA-9 offers several different anti-aliasing filters. The one used in the "Listen" mode at 192kHz has *perfect* response in the time domain - zero overshoot, zero undershoot, and zero ringing. However it is down about -0.5dB at 20 kHz. The primary filter used for MQA playback performs very similarly to that used in the Ayre QA-9. However they set a target of no more than -0.1dB droop at 20kHz. This requires a second digital filter which boosts the treble to compensate for the droop of their primary time-perfect filter.
This second filter introduces some "time blur", which is seen as the one negative-going undershoot you noted in your post. The concatenation of the two filters used by MQA yields a time response more like that of Ayre's "Listen" filter used at the 44kHz sample rate. This is inevitable, as there is no such thing as a "free lunch". It's simply a variation on the old story, "Price, performance, features - pick two." In the case of digital it becomes "Time response, frequency response, file size - pick two."
If one can tolerate the file size of quad-rate sampling, the errors in both time and frequency response are so small as to be negligible. With single-rate sampling, the file size is smaller but one has to choose between audible problems with either the time response or the frequency response. As Wadia showed us in the late 1980s, humans are more sensitive to time-domain errors than frequency-domain errors. Ayre has followed this path beginning with our first digital product nearly 20 years ago, and now MQA apparently concurs with this position. Hope this helps.
As always posts here are my own opinions and not necessarily those of my employer or slaves.
Many answers in link below-
John Atkinson, Stereiohile:
"I tried a variety of sample rates with these LP rips: 44.1kHz was very good, but didn't capture the essence of the original LPs' sounds; 96kHz was better; but there was no doubt that with a 192kHz sample rate I could not distinguish between the LP and the digital rip. And believe me, I tried."
.."there was no doubt that with a 192kHz sample rate I could not distinguish between the LP and the digital rip. And believe me, I tried."
So please, PLEASE tell my why need MQA again?
Charles, thanks again for addressing what so few seem to understand, and it is no wonder, with the confusing marketing MQA employs.
Yes, absolutely, they claim to correct errors at the time of digital capture by the original ADC.
But as you mentioned in previous posts, and as I know very well a project may have used MULTIPLE ADC's during the production. Then different ones during mixing and the same applies for mastering.
One thing I would like to point out with DSD, but specifically when remastering classic analog recordings:
There is a very good way to do it with NO additional DSD processing.
And that is do all your EQ and compression settinsg, and what ever
else in ANALOG, then capture that to DSD. Done. Essentially you are
taking a "DSD photo" of what is coming off the mastering chain.
That is how Mobile Fidelity and other audiophile labels are doing for their SACDs, and they sound spectacular to me.
The Pyrmaix workstation can edit and master in DSD, however some manipulations require conversion to so called "DXD" first.
Here is a thought, it may be crazy, but since your QA-9 is designed as you say, it would be VERY interesting to do some digital archiving with it and and then compare it an "MQA" processed version. The results would be very revealing, in a number of ways.
> > it would be VERY interesting to do some digital archiving with it and and then compare it an "MQA" processed version. The results would be very revealing, in a number of ways. < <
It seems to me that there is a big long chain that starts with the microphone in the recording venue and ends with the speakers in your listening venue. Improving anything at all along the chain improves the sound. Using a better sounding A/D converter would improve the sound for all customers, regardless of their playback system.
Unfortunately there don't seem to be as many "audiophiles" on the recording side as on the playback side. It isn't that easy to compare most pro equipment. Microphones are typically selected for known results ("desired colorations") in specific circumstances. Most gear isn't even compared - can you imagine trying to compare the sound quality of two different 128-channel mixing consoles? Occasionally a studio will re-wire everything with some "audiophile" cabling, but I'm unsure as to how they select which cable to use. It seems that endorsements from highly visible engineers count for as much as anything in the "pro" world.
Every time I have seen a "pro" audio guy evaluate gear, they will set it up in their system so that they can do rapid A/B switching. I have never seen anyone listen for more than 5 or 10 seconds before switching. This is very unfortunate as that type of listening test is extremely sensitive for only one subjective experience - frequency response. It is an exquisitely sensitive way to tell if two DUTs have different frequency responses. It is far, far less sensitive to other "audiophile" parameters such as imaging, soundstage depth, resolution, coherency, and so forth. And it is essentially useless for letting one know which DUT creates a greater sense of emotional connection with the recording artist.
I don't think there's much mystery about the sound quality of MQA. They always start with a high-resolution digital file. Then three processes are applied. One is to reduce the bit depth to ~17 bits with noise-shaped dither. This can never improve the perceived resolution. (NB: Various noise-shaping curves can introduce various sonic signatures. I am told that Sony/Philips listened to many choices before selecting the 7th-order filter used for DSD, and many have noted a similarity in the high-frequency characteristics of all DSD recordings, where the highs tend to sound soft, delicate, and airy, regardless of the program material - I remember having phono cartridges with similar colorations.)
The second process compresses quad-rate audio with lossy techniques, discarding further information. The third process is to add a (digital) filter to filter out some of the artifacts of the original A/D converter. This is also removing information contained in the original file. (It is easy enough to do this without the need for a proprietary system, as first demonstrated by Wadia in the late 1980s.) What sort of differences would you expect to hear from the type of processing that MQA performs?
As always, my posts are strictly my personal opinions and not necessarily those of my employer or voice coach.
"What sort of differences would you expect to hear from the type of processing that MQA performs?"
As you note, all of this can be done without a proprietary "format", god knows I use term oh so loosely, fees, and the need for new software and hard ware.
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