You missed my point. I'm referring to the ballyhooed improved sound of MQA compared to the traditional hi-res PCM mastering, perhaps (according to Bob Stuart) due to trying to restore phase and timing information that supposedly gets lost in the standard A>D and then D>A conversion, only restorable (again supposedly) by the complete end-to-end MQA system. As Mike L. posted, HQ Player is perhaps one alternate pathway to this goal. I think it's ultimately what PS Audio is working toward with their Direct Stream DAC's. As I posted, it seems unlikely to me that there is only one way (and a proprietary way at that) to achieve that sonic goal. I'm pretty sure the compression aspect of MQA is irrelevant to audiophilesIt is not possible unfortunately. Once you truncate the bit depth and sample rate in encoding for CD, nothing can restore it. It is a one way function that eliminates whatever was there. Many times there is nothing useful there so no harm done. But if there is something, conversion to CD format gets rid of it for good.
MQA restores that portion albeit using a lossy algorithm. And that process requires knowledge of the original high-resolution source so it must be end to end. Ditto for the conjugate filter they use.
DAC companies should use this as an opportunity to sell all new hardware than how to retrofit the old. That should put a smile on their face if they think about it.
Will MQA drive hdtracks, acoustic sounds, and other hirez sites out of business?
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I can't found information, that MQA someting do in time domain.
Official site and famous me patents contains nothing by this issue.
Here described some "MQA compliant filters and dithering" but nothing about timing https://mytekdigital.com/brooklyn-adc/#specifications
Also there nothing described: there is phase-distortion-of-DAC correction or not.
1. If it is possibly, could you give link where Bob Stuart says about it?
2. What is timing? It is jitter, clock unstability?
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You keep asking these same questions, perhaps you need to try to ask Meridian directly? In any case, here are some second-hand sources
http://www.soundonsound.com/techniques/mqa-time-domain-accuracy-digital-audio-quality
http://archimago.blogspot.com/2017/02/musings-discussion-on-mqa-filter-and.html
http://www.stereophile.com/content/mqa-questions-and-answers#w0efdqwwXMmFvHu4.97
Thank you for links. I can't ask Meridian about things which they don't claim.
In the links wrote about ringing only: "lazy" MQA filter vs. usual "brickwall". "Lazy" filter have lesser ringing tail.
If I understand correctly:
There suggested use such filter in each part of recording-playback workflow.
Traditional high resolution allow use "lazy" filter due band reserve too. But MQA allow save space at disk or band in broadcast stream.
Of course, minimizing ringing is one of aims of optimization of parameters during design of filter.
But, in my opinion, ringing is artifical feature, that suitable for technical analizys primarily. Because in musical signal almost no sharp changes between next samples.
May be I missed something (fast read with my bad English), but there no information about compensation of phase response and/or jittering in ADC/DAC.
Well, you have to remember that Mr. Stuart is acting as a salesman here, and like most salesmen he'll try to tell you enough to buy his product but not nearly enough for you to make that product yourself (so you need to buy it from him if you want it). Nevertheless, I see the same points being made (by him). 1) you need the whole process - MQA encoding after the ADC, then MQA decoding in the DAC, 2) a minimum-phase filter (characteristics not exactly specified LOL) is important and 3) eliminating errors in the time-domain (as opposed to the frequency domain, where attention has usually been focused, according to Mr. Stuart) is of major importance in accurately reproducing the sound on the master recording.
A couple of other points. First, it's a proprietary process, and it's in Meridian's interest (although not at all in our interest) to tell us as little as possible while still convincing us to buy into it (and buy it). Second, the process is applied to existing digital master recordings (so far, mostly but not exclusively, to recordings originally recorded on analog tape and then converted to hi-res PCM digital), so this all begs the question of why other engineers wouldn't be able to design DAC's which could accomplish the same thing (or better?) without using MQA?
A couple of other points. First, it's a proprietary process, and it's in Meridian's interest (although not at all in our interest) to tell us as little as possible while still convincing us to buy into it (and buy it). Second, the process is applied to existing digital master recordings (so far, mostly but not exclusively, to recordings originally recorded on analog tape and then converted to hi-res PCM digital), so this all begs the question of why other engineers wouldn't be able to design DAC's which could accomplish the same thing (or better?) without using MQA?
How to build "lazy" filter is not secret. It may be applied in both ADC and DAC.
In my opinion, filter may begin stop band since 20 kHz and end at half sample rate.
Some filter designers allow stop band end behind half sample rate.
If it applied in high resolution aliases in ultrasound are not listenable.
"Unnatural" pre-ringing is not objective feature. Minimal phase filter shift its energy behing the input front. And behing the front post-ringing with double energy.
I even don't know what option is better: "unnatural" pre-ringing or 2 times more distortions by minimal phase filter.
"Lazy" filter is good decission if signal and sample rate allow wide stop band.
I think, what "lazy" of "enough lazy" filter is not good decission for 44/48 kHz and for DSD64.
It is not and that is the answer I was giving. No interpolator or resampler (to any format) can restore what was truncated when the high-res file was downsampled to 44.1 Khz for CD. All that high resolution information is gone forever. HQPlayer cannot restore that. Or else we could use a 1 megapixel camera and have it be the same as a 16 megapixel camera. See this article I just wrote on audio and video resampling.
It doesn't have to be proprietary but does need to be end to end. As such it is a herculean task which few in the industry are capable of it. For that effort, they like to get paid for it.
Someone else wants to do this for free, should step up now but I see no one raising their hand.
It is not and that is the answer I was giving. No interpolator or resampler (to any format) can restore what was truncated when the high-res file was downsampled to 44.1 Khz for CD. All that high resolution information is gone forever. HQPlayer cannot restore that. Or else we could use a 1 megapixel camera and have it be the same as a 16 megapixel camera. See this article I just wrote on audio and video resampling.
It doesn't have to be proprietary but does need to be end to end. As such it is a herculean task which few in the industry are capable of it. For that effort, they like to get paid for it.
Someone else wants to do this for free, should step up now but I see no one raising their hand.
Amir, for a supposedly smart guy you can be pretty dense at understanding what others are posting. I think my post was pretty clearly NOT discussing a 16/44.1 file, but rather the original 24/96 or 24/192 or even 24/352.8 "master". I have to assume that the lossy compression part of MQA has nothing to do with its better sound quality compared to the original hi-res file (played back on non-MQA equipment), but you persist in ignoring that crucial bit of information.
And from what Bob Stuart has actually revealed about the way MQA works (see Yuri's posts above), there is no reason it has to be end-to-end, once you get past the fact that the lossy compression algorithm is irrelevant to that part of the discussion (i.e., improved sound quality).
Absolutelly for full band.
For downsampling 192 to 44 band 0...22 kHz is kept. After restoring 0 ... 22 kHz band restored back, 22 ... 96 kHz contains aliases (in ideal - nothing). In 0 ...22 kHz are loses due resampling, of course.
Yes. Any format conversion can't improve sound due new format only. Even if loses inaudible.
If DAC process (additionally decode) signal, it can have different distortions in two modes: MQA (additionally decoded) and PCM, that is don't decoded. So it is technically incorrect comparison.
Looks like you have lost track of your own question. This is what you said:
MQA is able to reduce the bandwidth of high resolution content down to CD's 44.1 Khz (encoding) and then restore what is taken out (decoding). No system that starts with 44.1 Khz at playback can do this.
The other feature of MQA is conjugate filtering which by definition requires analyzing the ADC filter used.
So on both fronts the answer is no. HQPlayer that you and Mike keep mentioning has no ability in any of this.
Then you don't understand what MQA is.
Hi Yuri. It is outlined in this paper (more or less): http://www.aes.org/e-lib/browse.cfm?elib=17501
Hi, Amir.
Thank you for the link.
There mentioned "triange" filter with 3 taps (3th order) that may be used in high band audio systems. It is filter with minimized ringing. But it is one of cases of traditional FIR filters.
While I can't understand: what in the article covered by patent (see part "6. PATENT NOTICE")?
Unfortunatelly, in the article no reference to patent that the technology are covered.
Hi Yuri. I have not read the patents. I think there are two of them and this is one of those: https://patentscope.wipo.int/search...=&sortOption=&queryString=&tab=PCTDescription
Explain to me how or why the lossy compression of MQA is necessary for the improved sound quality. I'll post this again, you seem to be missing it; audiophiles don't care what the file size is, they care about the sound quality. As far as "analyzing the ADC filter used", when this was first mentioned I thought that could be a real breakthrough. However, it turns out that that "analysis" is not done by any new or unusual measurement techniques, but rather just by listening!! So yes, I do think it does not need to be an end-to-end process, and I do think there is significant potential for other designers to equal or surpass the purported sound quality benefits of MQA.
I realize that MQA is designed to be an end-to-end process, and needs to be implemented that way for the listener to obtain its full benefits. However, that doesn't mean that a different design philosphy with the same goals re: sound quality has to be an end-to-end system, especially if it doesn't bother with lossy compression as an intermediate step in the process.
I am not aware of people praising MQA as compared to the original high-res content. They are praising it in the only context available: Tidal at CD rates versus Tidal with MQA.
The conjugate filter in theory can improve even high-res content but that is not the type of MQA content available.
So?
If you aren't aware of people praising MQA's sound quality compared to the original hi-res content, you have been reading only a small fraction (~10%?) of Internet posts about MQA's sound quality, and apparently none of the "professional" audio reviewers' comments in print or on the Internet.
The point about the ADC filters is simple (I would think); who is to say that MQA's engineers do a better (or even as good a) job at this than some other audio engineer(s) could? It's not as if they have developed a new or better analysis tool, and by now it should be apparent to anyone that what sounds "better" to one critical listener may sound "worse" to another, and vice-versa.
The point about the ADC filters is simple (I would think); who is to say that MQA's engineers do a better (or even as good a) job at this than some other audio engineer(s) could? It's not as if they have developed a new or better analysis tool, and by now it should be apparent to anyone that what sounds "better" to one critical listener may sound "worse" to another, and vice-versa.
Thank you, Amir. Looks like it is that patent. There date as 2014 year. But I can't understand what new (on 2014) in this patent for "lazy" filters. Coefficents of the filter? There claim contains filter with 6 taps, as I understand.
Principle of end of stop band upper half sample rate for lesser ringing used by famous oversamplers many years.
I suppose, as patent claim there considered full compression system (not filter as filter) 192 kHz (or other) signal in lower sample rate with minimization of loses via "lazy" filter.
Also interesting (in claims) what, if combine impulse responses encoder and decoder then total system impulse response will more compact than encoder's impulse response.
I think, if we record analog signal in 192 kHz and convert back to analog in a DAC, this system have no sense. Because in this case we don't compress-decompress/down-upsample digital signal.
As we have noticed before, Yuri, that patent has a lot of words describing the compression system, somewhat less space with the mathematics involved, and only a few words and no technical description to speak of regarding the filtering and subsequent minimizing of "time domain errors", or "pre-ringing" vs "post-ringing" in the anti-aliasing filters
And yet, virtually 100% of "professional" reviewers, and a lot of other Internet posters claim that this "system" results in better sound than just playing the straight PCM hi-res files (24/192, 24/96, 24/88.2, 24/48 or 24/44.1, whatever format the master is in)
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