The point of the MQA design was to provide an end-to-end, controlled and minimalist pathway. Many DACs have significant design issues and the idea was to correct some of them while controlling the filter paths. If you don’t have an MQA DAC, you just get the standard PCM path that your DAC uses anyway for everything else.
You seem to be arguing that there is something wrong with trying to provide a whole system, perfectionist approach. It’s okay if people elect not to buy an appropriate DAC, music will still play just fine, right?
But in a hypothetical scenario where the option for such a standard, lossless “PCM path” is no longer there, or is clearly stated as “less than great”, what do you do?
The handicap is clear for those not buying into MQA’s imposed vertical integration (not my problem, I insist - my Technics DAC/amp is fully on the MQA camp). Not even DSD has similar requirements, despite being trademarked.
Converter or filter errors on both sides can be corrected out of a chain but are not done so in current PCM or DSD processing.
The ‘lossless’ concept is fraught. It’s a seductive idea that everyone inherently believes, because it seems to promise no change to the signal after recording. The problem is that there is no reference for ‘lossless’. 96kHz/24b has no meaning either as an engineering reference or a psychoacoustic reference; it was never designed. The choice of bits and frequency happened historically as the quickest way to higher bits/BW without disturbing the system clock and staying on a byte boundary since digital is always byte oriented. Now consider for yourself that 144dB dynamic range (24b) is neither reproducible nor audible. The threshold of pain is 122 dB. And audio rolls off at a 1/f rate in amplitude. Moreover, 192/24 or 96/24 is only a release format; we usually record and process at higher sample rates and bit depths for processing headroom in mastering. So if you were to design a high res release waveform that accommodates what you think is audible with some safety margin, but doesn’t waste excess space on noise, what would it be?
As an aside, that’s an efficiency problem Bob Stuart has been publishing on for the last 30 years.
Of the mainstream artists I spoken to in the last couple of years studios use 24/48 PCM and are slowly moving to 24/96 PCM the releases you are seeing are probably from these PCM masters which are signed off by the artist or production team with no thought of MQA as it’s just not used or is in use in the process.
Plus multiple parts of demo’s, home mixes and studio recordings are all thrown into the DAW’s melting pot so you have no starting point to attempt to fingerprint the original sounds hardware components so no doubt it’s a best guest using one of the 16 pre defined filters available
The MQA encoder analyzes the given final signal, and the filter choices are tailored based on the analysis. They’ve published some of the aggregate data from the encoding of hundreds of thousands of tracks just as a way to document the range of noise and signal amplitudes found in recordings, old and new.
The white glove section of the MQA blog says they’ve also gathered a lot of data on A/Ds used over the last multiple decades. I don’t think there are that many A/Ds, tape machines, etc so it’s probably less an impossible undertaking than it sounds.
No, it analyzes the track it’s trying to encode and uses the noise and signal amplitudes (at least those two things, I’m not privy to the process), and chooses filters accordingly.
The published data is just a useful historic record to show how much variation in noise and signal has occurred as recording has developed.
The A/D data spans decades though, and that’s to enable the correction they are applying.
Yes, that’s the idea. I think the wide range of filter choices (are there 32 if I remember?) are to accommodate the various signal types they’ve encountered.