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Is MQA in your Future?

what baffles me is that when LPs are recorded to CD, they are better sounding than their CD-issue counterparts.

There are two possibilities. One is that they are mastered differently and you happen to not like the mastering job done for the CD. The alternative is that you like the added distortion inherent in LPs, and you get that when you record the LP onto a CD. If you want to read about the distortion, here is a good source:

Parts 1-5:


Parts 6-9:


It is worth mentioning that the measurements they are dealing with involve using high end equipment. Greater levels of distortion will occur with lesser turntables or if the turntable is set up improperly.


Back when digital recording began, but before CDs came out, audiophiles regarded digital recording as next best after "direct to disc" recordings. They were still buying their copies on LPs, so they got all the added warmth that LPs provide. It is only when people got digital material into their homes that some people complained about the sound of digital recordings. Evidently, they missed the added distortion that LPs provide, that give it extra "warmth."
 
Then it's not a real test! Per ASR!
I can give you measurements if you've an interest in what's happening below audibility for some reason, though measurements aren't necessary to demonstrate a point, and the basic physics of a null are typically more commonly understood.
 
There are two possibilities. One is that they are mastered differently and you happen to not like the mastering job done for the CD. The alternative is that you like the added distortion inherent in LPs, and you get that when you record the LP onto a CD. If you want to read about the distortion, here is a good source:
Good stuff... I ran my rig on Tube Amps for a long time, but then began to hear the level of distortion at higher volumes. I went back to SS and determined I liked the detail better. So yeah, I get ya... we like what we like!

And that brings me back to MQA. I've only heard it once at a friends house. He had a Tidal track in MQA and the same track as a digital file and on an LP. We listened to both, over and over. I preferred the LP track to the CD track because of what I percieved to be the level of digital hash. MQA blew both away with a level of expansiveness of the sound stage and inner detail! Was it all due to the time domain synch? God only knows, all I know is it sounded really good in my very limited experience.

So is there a future in a technology that can produce these types of experiences? I guess it's a matter of how much it costs!
 
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I can give you measurements if you've an interest in what's happening below audibility for some reason, though measurements aren't necessary to demonstrate a point, and the basic physics of a null are typically more commonly understood.
What did you use to create your null? A test tone? Was it as complex as my avatar? Lot's a variables there! You've obviously have your opinion, and I have mine!
 
What did you use to create your null? A test tone? Was it as complex as my avatar? Lot's a variables there! You've obviously have your opinion, and I have mine!

It's an excerpt from a track off of this album, though I can't remember which one. Not my kind of music, but it was an example track given by someone claiming that higher bit depth resulted in better base. It doesn't matter what the audio is- test tones or the most "complex" music you can think of, the result will be the same.

And it's not my opinion, it's simply how it works. You can use Audacity to do the same thing yourself.
 
Wow, interesting! So, it is taking the output of a phono-stage and feeding it through an ADC. Looks like you can just go Digital S/PDIF out to the DAC of your choice. I don't have an LP collection, but a good friend of mine does. I may play this by him to see if he wants to digitize some of his collection.

Thanks Tommy! Good stuff!
It's not difficult to do and the results can be quite satisfactory. I've "digitized" a number of my favorite LP's. The only issue for me is that it's time consuming. I've found that 16 bits @ 48KHz is more than adequate.
Then it's not a real test! Per ASR!
Oh yeah. ASR, the place where they review gear they often don't bother listening to. They carry very little weight with me.
 
He had a Tidal track in MQA and the same track as a digital file and on an LP.
Were these the same mixes?
Was the playback of each at the same level?
Was the playback of the digital sources via the same DAC?
Was the digital file of the same resolution as the MQA file? I'm going to assume it was lossless.

Or was it just apples & oranges?
And that's ignoring the 'was it a blind test' question.

Personally, I myself think there is technical merit in MQA with regards synch-ing the time domain.
We've discussed this before. I fear you still misunderstand their technobabble...

 
Not my words, but if you reasearch and read (like I do), maybe you'll understand!

"Aliasing corrupts a linear time-invariant (LTI) system such that its response can no longer be trusted to be linear and time-invariant. If I were to rephrase things informally, I'd say that: When a signal is undersampled in the frequency domain or has inadequate frequency resolution (aka freq domain aliasing), its spectrum has overlapping tails."

Something like the attached. But what does this mean? Well again, not my words but:

"It means that it is no longer possible to correctly reconstruct the time-domain signal. The spectrum tails do not go to zero, but are folded back. This inversion of the tails are what we know as aliasing and manifest themselves in the time-domain signal as impulse responses (a short-duration time-domain signal) that mix with the original signal.

These mixtures are perceived as distortion/roughness/etc. Their constructive/destructive effect give you your "quasi-random" amplitude modulations. The magnitude & position of aliasing in the time domain is related to the degree of frequency domain undersampling.

Finally, regarding "time smearing", the impulsive (quickly rising) parts of a waveform contain many high-frequency harmonics that when undersampled in the frequency domain, fold back as described above. However, I wouldn't call this distortion smearing, in the transient-smearing, or pre-ringing sense."


For reference see: https://ccrma.stanford.edu/~jos/sasp/Example_2_Time_Domain.html

So to those who think that 44.1Khz is more than enough sampling, or that MQA is nothing but 'snake-oil'.... I say one thing! Keep an open mind!
 

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And it all comes down to 'ringing'....

"Now what about digital music? Timing is everything. The first CD’s and CD players had great specs. But the sound was just not right. It did not sound alive. Where was the hall ambiance? Timing. Jitter. Theoretically, a 44.1Khz was a high enough sampling rate to fool the ears into thinking that digital sound was actually analogue. But is was not, at least not in the beginning. There was little hall ambiance. Stereo imaging was not good. The sound seemed lifeless. What could go wrong? Bits are bits. We are just dealing with zero’s and one’s. Yes, but they had better be in the proper order. Time smearing occurs when they get out of sync so to speak.

High sampling rates, apparently, make it easier to keep the bits in line. The skeptics will ask why should the sampling rate go out to 96 kHz or even 192 kHz. We can only hear frequencies to 20 kHz at best. Yes, but we are not talking about frequency response. We are talking about filtering out unwanted sound so that we can hear the sound that we want reproduced. It is easier to do the filtering at higher sampling rates. Not only do we want to eliminate unwanted artifacts from the analogue to digital conversion, we also want the timing of the bits to be correct. Low noise and low jitter. See the chart below:"
 

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So to conclude:

"We are looking at tiny differences, measured in microseconds. The ears can hear this difference. An impulse signal is sent through the system. The 96 kHz sampling rate rings longer and is less controlled than the 192 kHz sampling rate. This is a difference in the performance of the filtering.

The best performance, however, comes via MQA (Master Quality Authenticated) invented by John Robert (Bob) Stuart of Meridian Audio. It passes the input signal through but then it stops it short. Much of the unwanted ringing is removed which obscures the timing of the digital signal. In digital audio, timing is everything!

More can be said about MQA. For now, suffice it to say that it has shown great promise in the converting an analogue through a digital time domain and then back to analogue. Our ears want to hear analogue with as little time smearing as possible. Could this be the future of digital audio? We shall see."


Yes we will, won't we!
 
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Posted this in another thread, but thought I'd include it here. Perhaps many have not heard of K2 Processing from JVC-Kenwood. I really hadn't until today!

But is the JVC-Kenwood K2 Technology where MQA began it's path? It's interesting that the K2 Technology addressed the Time Domain issue with physical media back in the early 1990's. MQA tried to do the same thing with a streamed digital file. I wonder how much overlap there is of their technolgies?

Signal processing at time domain​

Instead of using the frequency domain, which is applied in similar technologies that restore data lost during digitization using frequency data, K2HD Processing uses the time domain to perform signal processing. This method restores lost data by analyzing each peak in the time-varying waveform; it is a unique method that no other audio manufacturer has used in the past.

The sounds of many instruments are based on harmonics, much of which are lost during digitization. K2 Technology creats the artistʼs musical expressions and subtle timbre of different musical instruments by analyzing waveform signals based on the time axis. It is a technology that is only possible with K2, as it processes based on the time axis not the frequency domain.


 
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