Zen & the Art of A/D Conversion Page 2

I shouldn't have been surprised. When I interviewed Charley Hansen (who, sadly, passed away in November 2017) in 2012 about the design of the QA-9, he explained that "instead of using the normal low-pass Finite Impulse Response (FIR) filter to turn the output of the delta-sigma DAC into PCM, we use a moving-average filter. This doesn't just allow for improved transient response, but actually perfect transient response. . . . There is no pre-ringing, no post-ringing—no ringing whatsoever."

Sample-Rate Conversion
These results reveal that, with the exception of the dCS 904 with its F4 slow-rolloff filter and the Ayre QA-9 in Listen mode, the signal, after being brickwall-filtered during A/D conversion, has been significantly altered from the original. Transients are smeared out in time both before and after the event by the converter's ringing at its Nyquist frequency. With CDs made from original analog signals that have been sampled at 44.1kHz, this ringing will be at 22.05kHz and, as you can see later, will not be removed by the DAC's reconstruction filter. However, it is increasingly common for digital master recordings to be made at sampling rates higher than 44.1kHz. The CD master is prepared by using a sample-rate converter, and the A/D converter's ringing at its Nyquist frequency will be an octave or more above the CD's passband. So if the master is sampled at 96kHz, for example, won't the ADC's time smearing be eliminated?

To answer this question, I took the 96kHz pulse captured by the Ayre QA-9 in Listen mode and used the highest-quality sample-rate converter in BIAS Peak to downsample it to 44.1kHz. The result is shown in fig.12. Despite the original digital data having perfect time-domain behavior, the sample-rate converter's low-pass digital filter has introduced our old friend linear-phase, sinc-function, acausal ringing, this time at the new Nyquist frequency of 22.05kHz. Again, this temporal blur now becomes part of the signal reconstructed by the DAC (footnote 4).


Fig.12 Digital-domain impulse response captured by Ayre Acoustics QA-9, Listen filter, at 96kHz, downsampled to 44.1kHz.

This can be seen in fig.13, which shows the analog output of the Mytek Brooklyn, set to use its short MQA reconstruction filter, when fed the Ayre QA-9's 96kHz-sampled pulse downsampled to 44.1kHz. (I used the MQA filter, as its own impulse response has minimal ringing and won't confuse the result, footnote 5.) The rate conversion filter's pre- and post- ringing at 22.05kHz has been affected by the MQA filter's minimum-phase impulse response (fig.14) but has not been eliminated. However, it is fair to note that the MQA filter's slow rolloff will not significantly attenuate content at 22.05kHz. I therefore repeated the test using the Brooklyn's FR (Fast Rolloff) filter. This will attenuate the data's ringing at 22.05kHz, but as you can see from fig.15, it appears to have substituted its own acausal ringing. (Note that the digital data has already been band-limited to half the 44.1kHz sample rate, thus is a "legal" signal.) The implication of this result is that with a musical transient, ie, when there is silence then data, the continuous waveform is exactly reconstructed after the transient, but before the transient there will be the sinc-function pre-echo present in the reconstructed signal.


Fig.13 Impulse response at 44.1kHz from fig.12, converted to analog with Mytek HiFi Brooklyn using the latter's MQA upsampling filter.


Fig.14 Mytek HiFi Brooklyn, MQA upsampling filter, impulse response at 441.kHz.


Fig.15 Impulse response at 44.1kHz from fig.12, converted to analog with Mytek HiFi Brooklyn using the latter's FR reconstruction filter.

And many listeners report that there is an improvement with transient sounds. I have written, for example, that a very quiet tick in the original WAV file of a choral recording, the Portland State Chamber Choir performing Eric Whitacre's Water Night, sounded more like a sound made by a human being in a real space in the MQA version.

What This Means
With rare exceptions, the ringing at the sample rate's Nyquist frequency is ubiquitous with A/D converters. However, it's difficult to see why this should matter. While there will be some few listeners who hear a tone at 22.05kHz with CD data, no one will hear a 48kHz tone with data sampled at 96kHz, or a 96kHz tone with 192kHz data.

Keith Howard investigated the "energy smear" exhibited by digital filters in January 2006, writing that "the time-domain performance of anti-alias and reconstruction filters has increasingly been blamed for CD's residual failings." However, he concluded that the listening tests described in that article showed that the energy smear "seemed surprisingly reluctant to show its face"; only in one extreme case—a filter in which all the ringing occurred before the impulse—did this energy smear prove consistently audible in listening tests.

Nevertheless, I understand that the ear/brain acts as a detector of wavefront arrivals, and that the pre-ringing of an acausal digital filter causes confusion: the initial onset of the ringing and the arrival of the maximum energy peak are incorrectly interpreted as two separate events rather than as one—as implied by Bob Stuart in his June 2018 interview with Jim Austin.

This might be one reason listeners prefer the sound of recordings made with higher sample rates (footnote 6). With 96kHz data, the time delay between the beginning of the sinc-function envelope and the maximum energy peak will be less than half what it is with CD data, and with 192kHz data it will be less than one-fourth that duration. Each time the sample rate is doubled, the time delay—and thus the confusion—will be half what it was before.

But what can be done to eliminate this confusion?

To a large extent, designing a D/A converter with a reconstruction filter that preserves time-domain accuracy and is free from pre-echoes is a "solved" problem. Almost since the launch of the Compact Disc, DAC designers have developed reconstruction filters that better preserve transient information. Two early examples were Wadia's DigiMaster filter and Pioneer's Legato Linear filter, both from the end of the 1980s—and many modern D/A processors offer a choice between such slow-rolloff, "gentle" reconstruction filters and ones that constrain the signal more in the frequency domain, at the expense of the time domain. MQA's reconstruction filter, for example, is a slow-rolloff type intended to preserve the timing of transients.

The trade-off these "gentle" filters offer for the preservation of the time-domain performance is a slight rolloff in the top audio octave and the increased possibility of image energy leaking into the audioband. As Charley Hansen wrote in a posting to the Audio Asylum web forum in 2004, "A gentle filter . . . becomes a trade-off between high-frequency rolloff and out-of-band energy. Most manufacturers choose a compromise with some high frequency rolloff and some out-of-band energy. . . . Most music has very little energy close to the Nyquist frequency. . . . [A]ny theoretical problems with out-of-band energy are probably no worse than the out-of-band energy found on LPs when played back with MC cartridges."

But what if the D/A processor has no reconstruction filter at all? By definition with these non-oversampling (NOS) DACs, there will not be any new Nyquist-frequency ringing, which might be why some listeners prefer them to DACs with digital filters. However, the ultrasonic images of the audioband spectrum are there in full force with such DACs and this will be less of a problem with high-sample-rate data than with CD data, a point made by Hansen in his 2004 posting. But the presence of this high level of ultrasonic energy may well lead to slew-rate limiting, hence distortion, with preamplifiers and amplifiers. And NOS DACs faithfully reproduce the A/D converter's Nyquist ringing on the recording.

Before he passed away at the end of November 2017, Charley Hansen and I were having a long e-mail exchange about digital filters. He sent me some measurements showing the behavior of experimental complementary slow-rolloff filters that he and his team had developed for both A/D and D/A conversion. The goal was to optimize the digital chain's behavior in the time domain by using a very "short" antialiasing filter at the A/D conversion (like the Ayre QA-9's Listen filter), and a similarly "short" reconstruction filter when the digital data are decoded. "The double- and quad-rate filters," he wrote, "are essentially the converse of the double- and quad-rate filters on the QA-9." The measurements were convincing. An analog signal encoded with Ayre's Listen filter, then decoded with the experimental complementary filter, was reproduced with perfect time-domain performance. The tradeoffs in the frequency domain were a rolloff in the top audio octave that reached –4dB at 20kHz with a 96kHz sample rate, equivalent to 6m of air.

Charley Hansen was a vociferous critic of MQA. However, if you compare what I've written in the paragraph above with my earlier description of the goal of MQA's developers, they are, in essence, identical! The impulse encoded at 96kHz with the Ayre QA-9's Listen filter and decoded with the Mytek Brooklyn's upsampling MQA filter is reproduced with its time-domain behavior preserved (fig.16)—in other words, from analog original to analog re-creation, there is no temporal blur.


Fig.16 Impulse response captured by Ayre Acoustics QA-9, Listen filter, at 96kHz, converted to analog with Mytek HiFi Brooklyn using the latter's MQA reconstruction filter.

The QA-9's Listen filter is a special case. But if MQA's removal of "temporal blur" is equivalent to MQA having eliminated the Nyquist-frequency ringing introduced by the antialiasing and decimation filters of conventional A/D converters, then the end result will be as if all of the recordings encoded with MQA were made with the Ayre QA-9—a fitting if unexpected tribute to Charley Hansen.

Footnote 4: The exception is the upsampling "apodizing" reconstruction filter developed by Peter Craven and used in Meridian DACs, which has a null at the CD's Nyquist frequency.

Footnote 5: Although it is possible to set the Mytek Brooklyn to use its upsampling MQA filter with regular PCM data, that is not the same as one of the set of MQA reconstruction filters, which is controlled by the encoder. These MQA filters are complementary to the set of anti-aliasing filters used by the MQA encoder.

Footnote 6: Joshua D. Reiss, A Meta-Analysis of High Resolution Audio Perceptual Evaluation." JAES, June 2016, Vol.64 No.6, pp.364–379.

Bogolu Haranath's picture

"It's Time" ........... Imagine Dragons :-) .............

jeffhenning's picture

...in fact, usually, very little knowledge is what is exhibited by people who are dilettante A-holes with strong opinions about audio that disagree with you, John.

MQA is rather brilliant. While I understand why it’s not being universally adopted, most people who aren’t engineers who have strong opinions about it don’t even understand basic audio recording and engineering.

Give them the credence they deserve (none).

barrows's picture

What footnote 4 appears to indicate, is that all that is required to eliminate the "time smear" introduced by the A/D process is the use of an apodizing filter in the playback DAC. Problem solved without all the additional expense and nonsense of MQA. BTW, one can implement an apodizing filter by oversampling lower sample rate content via the DSP capabilities incorporated into ROON or the use of iZotope oversampling incorporated into Audirvana Plus, all in real time, during playback. All this shows there is no need for, or reason for MQA and its requirement for the audiophile to purchase new hardware with MQA. Just oversample in ROON, or implement an apodizing filter in iZotope using Audirvana Plus and you have solved the "time smear" problem yourself without the need for (purchasing a new DAC with) MQA.

Bogolu Haranath's picture

"Satori"? ...............
Zen leading to satori :-) ................

jimtavegia's picture

I find little to complain about these days with all the 2496 recordings I can make that sound superb with only average recording equipment and I can complain about nothing. Music reproduction has never been better, but why people would listen to MP3 quality in 2018 and beyond me is still a sad thing to us as music lovers. All I every wanted was a better turntable and cart., better speakers so I could hear more and now for all too many less is just fine. We will never get it.

vclements's picture

I would have to agree.
Quite honestly I didn't read the entire article as I am on the road, tired, and have been teaching tech stuff all day..so I didn't want to read more tech stuff.
However, you summation is right on the money. I am very happy with 24/96 and better recordings...and as a road warrior, I can take good sound on the road with me!

Graham Luke's picture

...to MP3 because much of it can sound exceedingly good AND I am a music lover. Does that make me some kind of 'heretic'...?

Jason Victor Serinus's picture

it certainly has the potential to limit your ultimate enjoyment and appreciation. Perhaps your equipment smoothes things over to a large extent?

Graham Luke's picture

I appreciate that the sound is not as good as my ALAC files, some of the classical 320 kbps files I have received from the Chandos on-line shop are so good that I can easily ignore their prevenance and just enjoy the music. Isn't that what it's all about...?

John Atkinson's picture
Graham Luke wrote:
I appreciate that the sound is not as good as my ALAC files

You can find my technical analysis of lossy formats at www.stereophile.com/features/308mp3cd/index.html.

Graham Luke wrote:
some of the classical 320 kbps files I have received from the Chandos on-line shop are so good that I can easily ignore their provenance and just enjoy the music.

MP3 does perform at its best at that bit rate, though Apple's AAC codec does better.

John Atkinson
Editor, Stereophile

Bogolu Haranath's picture

Does the "law of diminishing returns" play any role? .......... When and where does the law start? ........

Bogolu Haranath's picture

May be the problem is "timing errors" rather than resolution per se ........... May be they are inter-related? ..........

Bogolu Haranath's picture

If I remember correctly, Bob Stuart (MQA) said that 20 Bit data is sufficient resolution ........... I am not sure about, what sampling frequency he said was sufficient ........

ok's picture

sports tons of ambient and detail at 320 kbps, but highs are somewhat truncated and hard, presumably due to the pre-ringing effect inherent to the otherwise superior Ogg Vorbis codec..

foxhall's picture

I Agree about Vorbis being a great codec.

I'm running Spotify through Amarra sQ+ which, to my ears, reduces some of the hardness in the higher frequencies.

Graham Luke's picture

I rate much of Apple's AAC material very highly. 'Mastered for iTunes' files can sound fabbo. (I love your site; keep the good stuff rolling!)

jeffdyer's picture

In one post you mention sounding good, yet in the next you disregard this and mention enjoying the music.

I agree you can enjoy MP3 files of good music, even if they don't sound very good. But no MP3 at 320kbps is going to sound as good as a 24/96 FLAC. You might enjoy it as much, but in hifi terms, there are lots of issues.

Graham Luke's picture

...how I have contradicted myself at all. I enjoy listening to music first and foremost. As for the rest, well whatever floats your boat, eh...?

spyder1's picture

I make MP3 files for specific applications from 192,96,44.1 CD-Res, that are listenable.

ok's picture

in all things audiophile at the down of the 21th century? Well, fine-tuning and convenience aside, absolutely nothing whatsoever. Therefore MQA or something HAS to be fresh and exciting – somehow, anyhow..

Bogolu Haranath's picture

Analog vs digital ....... tubes vs solid-state ........ hi-res vs mp3 ......... you choose ........ whatever floats your boat ......... Enjoy the music ........

Jason Victor Serinus's picture

It seems as though everybody is commenting on everything other than the article at hand.

texanalog's picture

And that fact reveals a lot.

Archimago's picture


ok's picture

So how about pristine 44.1 red book with no low-pass filter whatsoever and merely letting our ears cut off ultrasonic aliases? Rumors of wideband amplifier oscillation have never been confirmed for that matter as far as I know. I presume file size/bitrate is not actually an issue in the days of real-time HD video streaming..

AJ's picture

...Their Unprocessed State as Performed by Three Groups — Expert Listeners

This paper aims to examine perceived clarity in MQA encoded audio files compared to their unprocessed state (96-kHz 24-bit). Utilizing a methodology initially proposed by the authors in a previous paper, this study aims to investigate any reported differences in clarity for three musical sources of varying genres. A double-blind test is conducted using three groups—expert listeners, musicians, and casual listeners—in a controlled environment using high-quality loudspeakers and headphones. The researchers were interested in comparing the responses of the three target groups and whether playback systems had any significant effect on listeners’ perception.
***Data shows that listeners were not able to significantly discriminate between MQA encoded files and the unprocessed original due to several interaction effects.***

Authors: Generale, Mariane; King, Richard; Martin, Denis
Affiliations: McGill University, Montreal, Quebec, Canada; The Centre for Interdisciplinary Research in Music Media and Technology, Montreal, Quebec, Canada
AES Convention: 144 (May 2018)

foxhall's picture

I think it's silly for anyone to question whether or not MQA is elegant or not and it IS a brilliant innovation in my mind. The issue, for me, is ultimately traction and the very difficult path to making it matter to anyone other than producers/engineers and the high end community.

Most in my social circle are lifelong music fanatics but have zero interest in something like MQA and there's no use trying to convince them they need new stuff when they are enjoying music so much as is (via streaming, local files or records).

Archimago's picture

Where's the "elegance" again in MQA?

Which part is the "brilliant innovation"?

How exactly is this lossy compression system "high resolution" even?

Presumably this is the last of Stereophile's series on MQA. Showing a few impulse responses "encapsulates" all there is to the claim of "deblurring", I suppose.

What else to say but time to move on.

Graham Luke's picture

...is in the marketing. Throw enough largely incomprehensible goobldygook at people and it HAS to be betterer.

Bogolu Haranath's picture

Surprising ......... Nobody mentioned or opined about Bluetooth :-) .........

John Atkinson's picture
Bogolu Haranath wrote:
Nobody mentioned or opined about Bluetooth :-)

You can find my technical examination of Bluetooth codecs in my measurements of Chord's Chordette Gem DAC: www.stereophile.com/content/chord-chordette-gem-da-processor-measurements.

John Atkinson
Editor, Stereophile

Bogolu Haranath's picture

Thanks JA .......... I read the article .......... The article is from 2011 (7 years old) ........ I know you (JA) reviewed Chord Mojo .......... Any interest in reviewing the new Mojo/Poly combination? ...... may be other newer Bluetooth capable DACs? ......... One suggestion is Arcam irDACII ...........

John Atkinson's picture
Bogolu Haranath wrote:
The article is from 2011 (7 years old)...

Yes, but the Bluetooth codecs have not changed. Their performance is limited by the low bit rate.

Bogolu Haranath wrote:
Any interest in reviewing the new Mojo/Poly combination? ...may be other newer Bluetooth capable DACs?

No, because their performance cannot compensate for the limitations of the Bluetooth codecs.

Bogolu Haranath wrote:
One suggestion is Arcam irDACII ....

I reviewed the irDAC II, in July 2017 - see www.stereophile.com/content/arcam-irdac-ii-da-processor - including measuring its performance via Bluetooth.

John Atkinson
Editor, Stereophile

1audio's picture

Bob Stuart's comments leave some questions still to be resolved. Even so just looking at the electrical performance of the digital chain is overlooking other real system constraints.

First, real-world recordings made with real microphones have intrinsic low pass filters with the attendant phase response in front of any recording system. I don't know of any recording microphones with useful response to 60 KHz or anywhere near that. Measuring mikes that can respond that high have way too much noise. Plus the short wavelengths make every physical element near the mike have an impact where the longer wavelengths are much less perturbed. Same physics for recording mikes. Some have 1/2” diaphragms but the most desirable ones have 1” diaphragms and are hard pressed to get to 20 Khz. Still it would be valid to add (convolve) the impulse response of a microphone that has that response to 50 KHz to the impulse response of the digital recording chain and see what you get. Same for a Neumann U47.

I would really like to see some real impulse and tone burst measurements through the complete MQA system.

The concept of conjugate filters and dynamic sync of the encode/decode parts of the chain were first introduced by Keith Johnson and Pflash Pflaumer in the HDCD system. Along with encoding a marker to flag the content as HDCD. It has many of the same features as MQA and was compatible with 16 bit CD. However, it never had real marketing clout.

Anton's picture

Any chance you could compare those impulses as reproduced by the speaker of your choosing?

Bogolu Haranath's picture

About microphones ............ Some of the readers may not be familiar with Ehrlund triangular diaphragm microphones ......... Search for Ehrlund microphones on Google .......... Ehrlund claims that their triangular diaphragm microphones are superior compared to round diaphragm microphones because they have better impulse response ........... hence, low latency, less time smearing, low IMD ......... Very desirable properties ...........
Since JA is a recording engineer, he may be interested in these microphones ........... JA may be interested in making some recordings with these microphones .............

TonyF16's picture

How can Stereophile see MQA as the solution to poor digital recording/replay sound quality? All these words, but nowhere any reason for pushing another lossy packing app. when the world does not need one. Lossless is available already, has been available for a long time, and data rates + storage are no issues in our time for those who want lossless quality. Lossy is ubiquitous, and has been for a long time - for people who want to pack a lot of music on to their portable devices, at the expense of some compromise of quality. MQA has been around quite a while now and has made close to zero impression anywhere except in Stereophile.

I was pleased to read that you acknowledge that Charley Hansen (i) had his head screwed on and (ii) was a strong critic of MQA. Charley and I exchanged a lot of emails and he was indeed very knowledgable and sensible, and delivered fine products. However it is disrespectful to his memory to say: "Charley Hansen was a vociferous critic of MQA. However, if you compare what I've written in the paragraph above with my earlier description of the goal of MQA's developers, they are, in essence, identical!"

The conclusion is idle talk when it comes to justifying MQA as a finished product rather than talking about the goal of its designers. Donald Trump has "goals" of making America great again and to establish world peace. Great "goals" few would choose to take issue with - but the practicalities end up less easy than the statement of a "goal". MQA is fundamentally a lossy packing app., albeit a clever one which should sound better than mp3 - but nonetheless lossy and consequently fundamentally flawed. Where is Stereophile's listening test evaluating the losses MQA imposes on a lossless original recording? Has anyone independent of MQA Ltd. done any such test? If you subtract the MQA'd version of a 192k/24 source file from the lossless original, what do you hear? The big question remains - who needs MQA lossy packing if the pursuit is audiophile transparency?

When will this dead parrot be laid to rest?

CG's picture

There's some observations about the various digital filters over here: https://www.stereophile.com/content/kalista-dreamplay-one-cd-player

Charley Hansen also pointed out, more than once, that the digital filter is an important part of the system equation, but was well down the list of what affects the perceived sound quality.

steve21's picture

I've not been able to hear an A / B demonstration with MQA and non MQA digital recordings so I have no opinion yet. However any format that can compress 24/96 so you can stream it and replace 16/44 bit would be welcome. For me 24/96 is the sweet point, IF you can get it. Too many on line resellers just give you upsampled 16 bit. Until then i'm all about working on my music collection and kinda believe in the Mike Moffat thinking on music collections. Buy gear for the music you have. Regardless, if I listen to vinyl or digital or streamed digital, I find the original recording and mastering quality is the absolute pivotal factor in listening enjoyment.

CG's picture

I agree with you. (I bet that completes your weekend!!)

Between all this transparency talk is a basic fact that's hard to get around. That is, except for a very select group of recordings that were made with two microphones and were not edited - except for cutting into "tracks" - every single recording out there has been deliberately modified.

Somebody, or a whole chain of somebodies, has mucked with the mix, boosted this and cut that, equalized to their taste, and added various forms of compression.

Sometimes it was done to make the track more "catchy" when played on the radio.

Sometimes it was done so that the record could be played on kiddie phonographs (see Fremer's recent comments on Led Zeppelin II).

Sometimes because somebody thought it sounded cool. (Think of all those pre 80's recordings that were "digitized" through somewhat crappy A-D converters by some young recording engineer who was fed a diet of coke - the powder, not the cola drink - to help him stay awake enough to keep the march to CD going as fast as possible)

In any case, it's hardly transparent to the performance. If, in fact, the performance was actually a bunch of musicians who assembled into a room to play together. Very often, that's not the case.

So, in turn, it's hardly unreasonable for a home listener to tailor his or her system to play his/her favorite music the way they want it to sound. Why not? They're doing the listening. It's their home.

We all tend to get worked up over some detail of a single component in the overall system. In truth, barely any attention is paid to the electrical interaction between the various boxes we all use. Heck, most people are unwilling or even unable to improve the acoustics of the room they listen within. (This is an area the headphone guys really have it knocked.)

Slightly OT: This was really driven home to me just before the turn of the century/millenium. My daughter was going to play her guitar in the spring concert at her elementary school.

So, I decided to record it.

We headed off to the local Nobody Beats The Wiz to buy a Sony Walkman Pro. I purchased some wear-on-your-eyeglasses microphones from Sonic Studios. I sat in the back of the room and sat stock still while recording this.

Later, when playing it back on the home system, I was completely amazed at how close to the real thing it sounded. I never, ever, expected that. We would have been thrilled just to have the Instamatic equivalent of the concert sound. But, it was much, much better. Far more realistic in terms of what the original sounded like than virtually all our commercial recordings. At least what we imagine a bunch of musicians playing in a room would sound like. (My wife is actually a degreed musician who had studied with some of the people we have recordings of, so I have to take her word for it. I am just a guy off the street.)

This was all done by a guy (off the street...) who didn't know what he was doing with equipment that hardly qualified as professional quality. OK, I guess the Pro in Walkman Pro suggested professional, but I don't think many recording studios based their work around a little cassette recorder in their control room.

What conclusion do you draw?

ok's picture


Stevens's picture

Are people still seriously talking about MQA?
When it was first announced, the USP was enabling HD streaming at standard definition bit rates. That is now firmly ancient history, I've been streaming Qobuz up to 24/192 for several years and all you lucky audiophiles in the USA will be able to do so when they launch Stateside in the next month or so.

Ever since it's been looking for a reason to exist and has yet to find one. At the most basic level, when the CD was brought to market in the early 1980s anyone who was semi-conscious with some residual hearing capacity could tell the difference between vinyl and CD. It did not matter whether the listener preferred one or the other and, bizarrely, that argument still whimpers on. All that matters is that there was a difference to justify the new format, besides a quantum change in practicality and durability. Most listeners of classical music left emotion out of the debate on entering the digital world and never looked back.

The fact that several blind listening tests have been carried out and that it has not been shown with statistical validity that MQA and non-MQA can be told apart should be enough to consign it to the dustbin of audio history. Reference above to "expert listeners" is enough to raise my blood pressure to a level at which the only release is to grab the largest kitchen knife available and murder the nearest sofa. Any mass-market product must have perceived benefit to the average punter (the man on the Clapham omnibus), and if your market is limited to expert listeners then the audio dustbin should be close at hand.

Which gets me to the article. Finnegans Wake was an easier read, made more sense and was funnier. I hope the article made sense some people, it made none to me, I thought someone was having fun publishing an article in Ethiopian Coptic. If MQA requires understanding at this level to convince people to adopt it, then I suggest Bob Stewart could spend his time better introducing ice hockey into Namibia. He'll make as much money as he'll make out of MQA (none), but will realise the error of his ways a lot quicker (due to the lack of ice).

On the subject of Namibia, I was there recently with my Mojo DAC, Mr Speakers headphones and about 50gb of offline Qobuz HD downloads on my phone. I was at one lodge with no services at all - water was filtered from a dry riverbed, solar electricity, occasional wifi as they'd installed a few repeaters, the nearest road was 2 hours away and the nearest mobile signal a further 3 or 4 hours. As I sat on a sand dune listening to Bach, all was peace and I couldn't have been happier. The sound was sublime. Try convincing me I need another format.

FWIW (as they say these days), my unit from Devialet turns anything that hits it, including analogue, to 24/192 and does it's jiggery-pokery on that, including BBC Radio 3 that is the bastion of broadcast technology and civilisation as we know it, and broadcasts at mp3 or lower.

leonardlaub's picture

Thanks, JA, and kudos for your fine article on A-to-D conversion in the September 2018 issue!

After all the widespread handwringing about evil commercial interests foisting an MQA monopoly on innocent music lovers, it's a relief and boon to see your careful, objective, fully substantiated demonstration of how:

- Typical digital audio recorders (from which come the hi-res PCM files many regard as the best digital audio can do) introduce time smear and ringing during A/D conversion which typical playback systems can't and don't fix, and to which those playback systems' D/A converters typically add;

- A/D converters following a particular unusual design approach don't introduce significant time smear or ringing during recording; and

- Recordings made that way can then show substantially perfect impulse responses if D/A conversion in playback is also done following a particular unusual design approach.

Your final paragraph then calmly pointed out that if MQA can indeed reach upstream into existing PCM files and remove the time smear and ringing caused by typical A/D converters during recording, then MQA encoding could permit those existing PCM files to be able to be played back with substantially perfect impulse response.

You neither endorsed nor dismissed MQA. You did show that substantially perfect impulse response can be attained in a digital audio system if both A/D and D/A operations follow specific but not typical approaches. You left room for MQA to demonstrate it can not only provide the necessary D/A conversion but also remove time smear in existing recordings not made with suitable A/D converters (i.e., nearly all, though not the ones you've made with the Ayre A/D, thank you!).

Those of us who especially value the time-domain performance of audio systems welcome the prospect of routinely hearing music reproduced with all attacks, rim shots, and other interesting transients delivered without smear (at least until the speakers make their contribution).

Please persevere in measuring and presenting impulse responses for digital gear, step responses for speakers, and fast square wave responses for analog electronics. We can hope manufacturers begin to recognize the value of improving all those.

Thanks again!

Currawong's picture

... has been solved. You design a DAC with a much higher tap count such as Chord and iFi have done. The problem is, we're still stuck in the past with DACs that haven't changed design in over a decade, with lousy built-in filters, leading MQA to come along and try and "solve" a problem that should have been solved by manufacturers. Unfortunately now the "solution" is to run all the music through some kind of compression so they can say "Listen! It sounds better!" which wouldn't have worked if it were just compressed high-res music. To me, the whole b-spline etc. discussion is just to avoid the fact that they can no longer say they are reproducing the studio master, as that has been shown, very comprehensively, not to be true.

Regardless, the discussion in this article on digital ranges, as I understand things, from out-of-date at the very least, to wrong. Infinite slope impulse responses, don't exist in music, as they contain out-of-bandwidth frequencies, so using the ringing caused by a DAC's response to out-of-bandwidth signals to explain what is going on is nonsense. Show me an in-bandwith signal that has ringing using a conventional filter! I'll bet that you can't!.

John Atkinson's picture
Currawong wrote:
Show me an in-bandwith signal that has ringing using a conventional filter! I'll bet that you can't!.

Fig.15 in this article shows just that. With the DAC decoding data restricted to "legal" frequencies but with sinc-function content at Nyquist (fig.12), the DAC's own filter rings. (Note that this reconstruction filter attenuates content at Nyquist by >20dB, so the ringing in this graph is not that in the original data, but is the filter's own ringing.)

John Atkinson
Editor, Stereophile

Currawong's picture

That signal is exactly at Nyquist though, and irrelevant to actual music. It doesn't change the fact that musical content doesn't show ringing.

John Atkinson's picture
Currawong wrote:
That signal is exactly at Nyquist though, and irrelevant to actual music. It doesn't change the fact that musical content doesn't show ringing.

The measurements in this article indicate that every musical transient in the master will be accompanied by sinc-fuction ringing at Nyquist. This will be either from the original A/D converter (if the recording was made at 44.1kHz), or from the sample-rate converter used to create the CD master. That ringing will excite the playback DAC's reconstruction filter, which will impose its own ringing on musical transients.

Yes, a surprising result.

John Atkinson
Editor, Stereophile