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In Reply to: RE: Thanks... posted by Charles Hansen on May 28, 2017 at 12:35:01
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?
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