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No it doesn't...
Really? Explain to me how it corrects timing while using less data. The idea is you are only capable of hearing up to what MQA gives you and the rest is superfluous. That it 'corrects' already tells you they are taking liberties with the data. There is also no correction to an original file without a marker. There is averaging, guessing and assumptions to eliminate those nasty useless bits. It's lossy and has to be. If you lossless compress a 24/192 file to where you've thrown away every extra zero, it will generally still be over twice as large as any MQA file which requires more processing and load on the playback device. What gets streamed on tidal is an even smaller version.
No, that's mixing, not mastering. Mastering never involves mixing tracks, it's adjusting levels, EQ and processing, but ranges from lots of all of that to none.
Hold it. There's nothing right about the above. Nothing. Their claim re:correction includes only the "temporal blurring" aspects of the ADC and DAC filters. That's all, and that is of questionable efficacy.
I read the other day that the Dolby royalties for audio in DVDs were expiring. I'm no expert in this area, but I know they pushed to get their technology written into the standards so they would get a piece of every DVD sold. I can see a company like Dolby buying out Meridian and rolling it into some of their current standards just to start the patent clock ticking and extend their control. It might not look like it's expensive to the end user, because it would be way below the surface, but it would still amount to an awful lot of money.
Exactly like Apple's iTunes store.
By the way, "louder" and "harsh sounding" are two quite different things. Harsh implies frequency response imbalances or distortion, which I doubt that a streaming service would be responsible for causing. Have you tried balancing the line level before comparing? It may be the exact same sound, just at a different volume level.
You're picking nits as reverb can be added to separate tracks etc. It's much the same and engineers tend to not differentiate like that. What do you think they are implying here. Lining up multiple adc's etc in a mix and timing it up. What introduces 'temporal blurring' other than multiples falling out of line due latency etc. I fully agree it of questionable efficacy and extremely vague. It's an artifact of the process they claim to be an advantage.
first they make claims about timing, not about absolute resolution or fidelity. because that indeed wouldn't stand. now for timing, the basic simple trick is using higher sample rate. as they then compare the MQA 48khz container to a 48khz PCM, magic, they have better timing information as in the container they have 96khz sample rate. it's a trick and at the same time it isn't.
another aspect they claim to improve and they do, is that they reduce ringing at the band limiting frequency nobody really cares about but them. they do that by keeping high sample rate and by going for what I imagine to be a mix of filters the same way many moderns DACs do. they then want to play that game at the ADC, while selecting the output format and filter type of albums to be released, and at the DAC. the end result is that, congrats the ringing is almost gone and they can show a Dirac pulse for everybody to misinterpret how it looks, for fidelity. :/ of course his comes at a price. for a general idea only, you can think of using only an analog EQ to low pass. when you mess with FR you also change phase, and vice versa. here the entire philosophy of MQA is "we go all in for the time domain!". bit depth, boom pay the price. frequency response, boom pay the price. but they aren't lying about that, they really take great care of one out of 2 variables to define the signal.
as for file size, and lossy, yes the format is in some ways lossy as soon as they apply dither or go for some lossy compression of the ultrasonic data as they seem to have the choice to do. but the file size of a 24/192 isn't evidence of it. because no MQA content will ever be 24/192. once extracted it will be more like 18/192 maybe, maybe less. so right there you save a good deal of space. lower resolution does that ^_^.
Less would be a good assumption.
It might be worth your while to pay close attention to those of us with actual hands-on experience in the recording industry. You know, the actual "sound engineers".
You have blended the mixing and mastering processes into one, but in the professional world they are separate. Referring to processes such as reverb applied on a per-track basis has nothing to do with the claims about temporal blurring made by MQA.
In any multitrack recording system all ADCs are clocked together via one of several master clock methods. They are lined up by default and of necessity, but if one ADC didn't follow the master clock it could never be corrected once in the mix, not by MQA or any other method. And that's not what MQA claims to fix either.
MQAs reference to temporal blurring relates to the temporal response of a single ADC channel caused primarily by the anti-aliasing input filter. That response can be seen as phase response, group delay, or impulse response which are all different windows on the same data. The inputs of a multitrack recording system are mostly identical, but even if they aren't, unless they're sharing at least part of the same signal with each other, and other channel timing differences are simply not an issue. What MQ a claims is a problem is the time response of individual anti-aliasing and reconstruction filters, which they claim to correct, and they claim makes an audible difference. There is, of course, no real evidence, and given the lack of information about the types of filters and how they were used in any complex recording system, this would seem to be an impossibility.
MQA makes no claim to be able to be able to undo mixing or mastering, they're only looking at impulse response of filters.
I am one of the people saying there will be a fee at encoding, streaming, decoding, and hardware... because that seems to be what's required for the full experience. Those assumptions are perfectly rational, and one of the fundamental problems they (me included) have with the format.
That is faulty reasoning. They could have a deal with Dragonfly in order to have a cheap DAC on the market to help promote the format.
What they choose to distribute is a different matter altogether.
Hold on a second, there's no software encoder from PCM? That's what they must mean by their direct "analogue to analogue" experience. ADC->DAC, no conversion. I imagine the excuse for this kind of lock down over the encoders is their "authenticated" experience, but that makes direct comparison to PCM impossible for anyone but big labels. It's not an acceptable recording format, it has no business being used as the sole ADC source. Yet PCM to MQA conversion is impossible without software encoders. We know they exist, the conversion of digital masters to Tidal has been discussed ad nauseam already, so how did Tidal get ahold of the MQA versions? An MQA software converter exists. It must exist. What they plan on doing with it, or if it will ever become publically available appears to be anyone's guess.
This is looking more and more like vaporware to me. Not even a bum format. Just vaporware.
Actually I think they can do this without altering the phase, and it is this that is new. Say you want a low pass filter. One pole with give you 90 degrees phase shift, and 45 at the 3dB point. You cannot fix that. If you want 2 pole you get double that. Unless you do one of the poles in the digital domain, and you do it effectively backwards in the time domain (which you can do if you have the file). Then the phase in the forward playback direction is reversed. Add this to the other filter and you get cancellation, zero phase shift, and a 2 pole 12dB per octave low pass. Magic. It is FILTFILT in Mathcad
Obviously MQA version is a bit more complex, as they are compensating for a bunch of different ADCs and DACs.
"we go all in for the time domain!" is inaccurate, because while they are trying to do better there than others have before, apodizing tended to damage the HF through slow extra roll-off in the pass band, so they can fix that too. I think it is time the time domain got some attention. Everyone seems obsessed with fixing the frequency response, which is fine, but all the f-ing about with EQ screws up the phase response as you say above, especially in the LF.
That doesn't require actual fees at each stage. See above for discussion on Dobly
Perhaps, but that would put it isn direct competion with their own Explorer II, which discounted is 129 GBP (apparently we can consider discounts to make our point, see above)
Sure but that wasn't my point. Dolby here is redundant, as people here are claiming MQA is redundant, even though Dolby does nothing here, and MQA makes the file "smaller and better"
Now I or you are confused. I was talking about a possible domestic ADC. So the pubic can play. What is the issue there?
You know not to whom you speak so don't be so sure about what I do or know. What their process does to anti aliasing filters would have the effect I'm speaking of in the overall signal processing. They're likely lining up peaks which would save bits and allow them to claim better timing or correction. We're not really in disagreement here but like you, I don't think their claim can be accomplished and is just some 'optimizing'. My opinion is that it affects more than just their claim. Whether you're mixing or mastering, you are still dithering and rendering files.
Look at the curves MQA supply showing how they are differentiating peak with more resolution than 192kHz 24b and you may see why some here feel you may not be well informed.
MQA throws away the MSB above 20kHz, not what you are saying
It's arguable that time recognition extends beyond what is achievable by normal hearing bandwidth and losing bit depth can have a similar theoretical effect. They make assumptions and play by their rules which allows them to make the claims they do and lose what they feel are insignificant bits. I know I'm reading in but there's only so many ways you can do what they claim.
That may be MQAs argument, and why they are going to all this trouble. Many here do not believe it to be true.
If it were the LSB. But MQA almost entirely loses the MSB in the ultrasonic frequencies.
Yes, as does everyone. 192kHz 24b PCM loses samples and bits compared to 384kHz 32b. At some point someone decides what is enough.