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MQA Software / Hardware Decode Etc... Questions


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Perhaps where we disagree is that Warner did not publish the provenance of the source, and that the final HDTracks (and Tidal MQA) product sample rate is not justified by the actual provenance. I'm not disputing Mr. Ludwig's expertise, I just think that a more honest final consumer sample rate is 16/44.1.

I think the same argument can be made for *many* of these "high res" remasters ;)

 

For example, is there any reason an analog tape recording from the 1940s should be released at 24/192 or DSD256? 99% of anything above 20kHz is just noise / tape hiss / artifacts from the recording and / or playback equipment - why wouldn't 16/44.1 be perfectly adequate for those, as well?

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I think the same argument can be made for *many* of these "high res" remasters ;)

 

For example, is there any reason an analog tape recording from the 1940s should be released at 24/192 or DSD256? 99% of anything above 20kHz is just noise / tape hiss / artifacts from the recording and / or playback equipment - why wouldn't 16/44.1 be perfectly adequate for those, as well?

 

In my professional realm, we typically scan film (even nitrate) at 4K and make all the consumer end products from that source. Will a 4K release of Apocalypse Now (for example) stand out from the HD release? Film grain becomes more visible from sources of that vintage and before and de-grain (the video equivalent of noise reduction) can address it, but at the cost of "fidelity".

 

The debate about whether audio sample rates above 16/44.1 are "necessary" for the consumer has been raging for over a decade now (perhaps longer). Entertaining any such discussion in this thread will likely derail it past the point of redemption. But please feel free to start a new thread. I'm always curious about "audiophile consensus" or the lack thereof.

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I think the same argument can be made for *many* of these "high res" remasters ;)

 

For example, is there any reason an analog tape recording from the 1940s should be released at 24/192 or DSD256? 99% of anything above 20kHz is just noise / tape hiss / artifacts from the recording and / or playback equipment - why wouldn't 16/44.1 be perfectly adequate for those, as well?

 

Because the "native" output of the machines they're run through comes out of a sigma-delta modulator. As you go lower in resolution, your A/D filtering starts running into a similar problem to the one NOS DACs have - the closer 2x gets to 40KHz, the harder it is for filtering to do a good job (i.e., behave well - cause little or no audible distortion - in both the frequency and time domains).

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

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Skepticism is always cool with me, but you seem way beyond "somewhat doubtful."

 

A book on your shelf means absolutely nothing to me. The author is a smart guy, but I'm very doubtful he's make the bold statements you're making here. Perhaps it's because there are repercussions for him to say things unlike the anonymity of an online forum where one can push any agenda he wants.

 

I'm not aware of Oppenheim ever making any statements directly about MQA or Bob Stuart. However, when Stuart makes claims in direct contradiction to what Oppenheim, whose credentials are at least as good as Stuart's, writes in his books, one cannot believe both. Fortunately, this is mathematics, so nobody needs to take anyone's word for anything. The established theorems covered by Oppenheim and many, many others all check out right back to the most basic fundamentals of mathematics. I've followed the proofs all the way myself (that's part of any undergrad programme in electronics). So no, there isn't room here for "benefit of the doubt."

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I'm not aware of Oppenheim ever making any statements directly about MQA or Bob Stuart. However, when Stuart makes claims in direct contradiction to what Oppenheim, whose credentials are at least as good as Stuart's, writes in his books, one cannot believe both. Fortunately, this is mathematics, so nobody needs to take anyone's word for anything. The established theorems covered by Oppenheim and many, many others all check out right back to the most basic fundamentals of mathematics. I've followed the proofs all the way myself (that's part of any undergrad programme in electronics). So no, there isn't room here for "benefit of the doubt."

 

Perhaps you could give me a quick example of where Bob and Oppenheim would disagree? Nothing too in depth that requires you to write page after page.

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Thanks for all that info. But did you miss the thrust of my point? If industry leading mastering engineers tell us that its ok for redbook to be played back on an analog desk, remastered, and then resampled at a higher rate, who are we to tell them its fake? I think Bob Ludwig's words were along the line of "the higher sample rate accurately captures the decisions made at the [re]mastering stage".

 

If an industry leader spouts nonsense, that probably means the entire industry is (mostly) nonsense.

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If an industry leader spouts nonsense, that probably means the entire industry is (mostly) nonsense.

 

Putting the audio through an analog console and outputting to a high sample rate changes the sound. You can't deny this. That's all that really needs to happen for someone to say it's a valid thing to do. Right-clicking and converting from 44.1 to 176.4 on the other hand may not benefit from this change in sound.

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When it comes to real-world implementations of digital systems, as the 19th century semanticist Alfred Korzybski opined "the map is not the territory..." When considering a sampled and quantized signal, everything is an approximation after quantization. Engineer or not, considering the subjective effect of leaky filters on time-varying audio signals is worth your time. Theoretically, time-varying audio is a superposition of ideal voltage magnitudes as sine waves. Practically, not so much. It isn't fair to fail to acknowledge that hearing mechanism are still not fully defined by the research, particularly with regard to how the brain processes signals for timing and phase. The math only defines linear systems for your application, not non-linear systems. Haven't we already had this discussion?

 

image.thumb.jpeg.a4a84e289e35c7e49a6d3042fc9b2a99.jpeg

 

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I remember Bob Ludwig stating that the Rolling Stones DSD transfers sounded identical to the master tapes. Funny how when something new comes out it sounds perfect until the next best thing comes out.

 

It also bears noting that what audio professionals hear in a high end studio environment is rarely analogous to what we hear at home. Oh, and also, those Rolling Stones SACDs didn't sell very well (are very collectible today) back in the day and were quickly discontinued.

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I've heard someone else say MQA has issues with Nyquist but I can't remember exactly what the issue is. I believe it has been addressed by Bob, but again I'm not sure.

 

The issue is inadequate filters resulting in high levels of aliasing. I've only seen Bob address this by furiously waving his hands while saying it doesn't matter because reasons.

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This is valid, and certainly has some quarter in audiophila, but it ultimately boils down to subjectivity, the place where all the nefarious things in audiophila live.

 

I'm guessing this is important not just to audiophiles but to the listening public as well, who often go for a sonic result through Beats headphones.

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OK. It seems that Bob is going for a sonic result and you'd like him to go for a technical result?

 

I prefer to have an accurate representation of the music as originally recorded. If I want a "sonic result" I can apply DSP effects myself. Why should Bob be the one to decide what sounds good?

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I prefer to have an accurate representation of the music as originally recorded. If I want a "sonic result" I can apply DSP effects myself. Why should Bob be the one to decide what sounds good?

 

Preference is important. Bob has made design decisions to provide what he considers the best sound, given certain constraints. I believe he is going for accuracy as his sonic result, while you are going for a theoretical perfect looking picture that doesn't exist int he real world.

 

Who knows. I don't want to defend or talk like I know what Bob thinks.

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I prefer to have an accurate representation of the music as originally recorded. If I want a "sonic result" I can apply DSP effects myself. Why should Bob be the one to decide what sounds good?

 

How sure are you that correct frequency response, arguably at the cost of some time domain distortion, causes your ear/brain to form an "accurate representation" of the original?

 

 

Here's an example where amplitude (brightness) and perhaps frequency (color, though I'm not sure) accuracy are demonstrably unnecessary to provide an "accurate representation:"

 

 

 

 

 

Does this mean we ought to throw out the math? Not hardly. But perhaps we can't be quite so definitive about what is necessary for our senses to create an accurate representation. (Leaving aside for the moment very real differences in perception among different individuals.)

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Does this mean we ought to throw out the math? Not hardly. But perhaps we can't be quite so definitive about what is necessary for our senses to create an accurate representation. (Leaving aside for the moment very real differences in perception among different individuals.)

 

I sincerely wish you would overtly advocate for this "middle path" you sometimes hint at...

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