Chord TT2 - use Tidal or Qobuz?

All recording methods alter the original - even recording at 512 DSD,

Here is what MQA does:

https://www.soundonsound.com/techniques/mqa-time-domain-accuracy-digital-audio-quality

Technically it is an interesting way to transmit modern recordings. Mostly the original performance these days is captured using some one-bit high sampling rate such as 10XDSD. However, MQA claims the audible information is contained in a triangle found in figure 7 of the above link.

To transmit it efficiently, you first run it through a slow roll-off 32-bit filter flat to 20k. It slowly rolls off to be about 10db down at 48k and keeps on reducing even higher frequencies. The exact filter used is determined by MQA when they analyse the recording.  Since it is all noise above 48khz, that is not a worry.   Rob Watts has a 50khz filter in all his DACs to get rid of high frequencies that can cause issues for later stages.  Hopefully, MQA has done blind listening tests to prove the filters used are inaudible.  Suppose you have a 192k recording and chuck away every second sample, then you get a 96k stream. But this chucking away has a consequence - information above 48khz is reflected into the 0-48khz region. However, because of the shallow filter, the level of this information lies below the noise floor.   The filter chosen by MQA ensures that.  Hence it will not make any audible difference. You can do the same for 384k to convert it to 192k and so on, all the way up to the high DSD rates it was recorded at. So all the audible information has been captured at 96k.

Now you can locate the noise floor and figure out how many bits you need to reproduce the triangle's in the link I gave.  Add a couple of bits to the safe side, then reduce it to those bits using dithering techniques. Look dithering up if it is not something you understand. From what I have read, it can be anything from about 15 to 18 bits. This means the stuff that has been reflected has been chopped off. It then, for compatibility reasons, adds and subtracts samples next to each other. Compress the removed information in the bits below what is chopped off, and if played back at 48k 24 bits, it will simply sound like noise. But to get the 96k back, use those bottom bits to recover the difference information, so you get the original back by adding and subtracting. In practice, MQA uses what is called a quadrature filter - but the principle is the same. We can use linear interpolation for example to approximate what was chucked away.  We can do better because we know the slow roll-off filter used so can figure out the best filter to upsample it back to some high bitrate. It's just an approximation of what was chucked away, but since it is all noise, who cares.

The idea is to introduce the least audible processing to get that very high bitrate recording delivered to the listener.  I think all this merging into the bottom bits is silly - transmit the 96k using Flac.

OK, that is what MQA is supposed to do. You usually use a sharp cutoff sinc filter to reduce the very high bitrate to say192k and transmit it. The MQA claim in doing that it introduces time smear. The shallow filter introduces negligible time smear. But the MQA people forget one thing - Shannons SamplingTheorem. If you take a bandlimited signal and upsample it using a sinc filter, then the bandlimited signal is reproduced precisely. That is what the DSD and Chord DAC's do. There is no need to try and recover an approximation of just noise especially if a 50khz filter is used. Only a blind listening test can determine which is best. But MQA certainly is a tricky way of doing it.

Note, however, when audio is made into MQA, what is supplied is usually the 48k, 96k, 192k or even DXD master rather than the original very high bitrate recording. That will likely have used a sinc that MQA claim causes time smear. When converted to MQA, that time smear will remain, so their claimed advantage of reducing time smear to really low levels does not hold. It would have to upsample it to some high frequency then rely on the Shannon Sampling Theorem.  After apply the MQA process. That is where the controversy starts. People like Rob Watts claim, unless you use his 1 million tap filter to get an exact reconstruction to 16 bits, then audible inaccuracies are introduced. Most engineers would claim with modern filters, such inaccuracies are inaudible. Only a blind listening test can tell. But measurements show Rob certainly has produced a perfect sync filter to at least 16 bits accuracy with his M-Scaler. However, he has never published the math showing you need a million taps that require a lot of processing power.

Just a personal opinion, I think MQA has gone in the wrong direction in trying to be compatible with existing standards. One can apply the slow roll-off filter and use a new compression method invented by Microsoft:
https://www.microsoft.com/en-us/research/wpcontent/uploads/2016/02/Malvar_DCC07.pdf

You chop off everything below 18 bits in the frequency domain and transmit at the lowest rate where all frequencies above it are zero after chopping off to 18 bits. Any loss of resolution in the frequency domain is much harder to detect than in the time domain. It will likely produce better quality and smaller files and capture the information in the small number of recordings that do not fit into the triangle.

The bottom line is when playing back MQA through a Chord DAC, especially with an M-Scaler do the first unfold.   That way you are getting the trickery used by MQA to band limit the signal that does the least damage to the signal.

Here is a link to the microsoft paper that works:

What streamer are you using? Also, are you using Roon?

Some streamers offer MQA support (e.g Lumin) and Roon can also Decode and Render MQA. 
 

Qobuz is better choice for Chord as you point out about MQA.  The primary reason you would use Tidal is if you prefer Tidal's library and some of the other functionality.  That being said, if you are a Tidal fan, Chord might not be the right choice for you in terms of DAC.