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MQA part the 3rd - t't't'timing...

This means: the displayed data are very correct.

I don't think anyone was claiming they were incorrect.

The point is that Bob's demo signal is 1) very unchallenging and 2) displayed at a scale that cannot reveal any detail. IOW, Bob shows ... nothing.
 
I propose that the music industry adopt a certificate that guarantees the consumers that they buy genuinely "matematically lossless" files when they shop for LPCM redbook or hi res files.

Is that before you insert it into an upsampling DAC which will then “invent” some data?
 
It is a laughing matter that in order to defend the unneeded distortion by filtering and origami folding and unfolding of the signal that is introduced by MQA, Mr_Sukebe and others keep pointing to existing "distortion" or flaws in the replay chain that is common to all music files played through them.
 
I don't remember "master provenance" demands made around SACD.

We were just happy we were getting better sound.

Well, at the time I had a number of conversations with people involved with developing DSD for SACD which *did* include that issue. This was early on when despite having started to release SACDs and players they were still altering the modulator and demodulator designs as they found problems. As IIRC Lipshitz pointed out at the time DSD has some inherent 'semi-chaotic' problems like the risk of lockup of idlers.

However in some ways the 'master provenance' issue was different when the *source recording* was being made in DSD. Once that's done you get the result warts-n-all if you used a poor choice of modulator.

(N.B. For DSD the terms 'modulator' and 'demodulator' were used in place of ADC and DAC.)
 
I have also in one case found the CD layer of an SACD dynamically compressed compared to the SACD layer.

Such shenanigans and our worries about them are not particular to MQA.

A complication there is that some discs are actually *HDCD* / SACD hybrids. So if you play the 'CD' without HDCD decoding the result may well be compressed. So check the text with care.
 
Well done 320k AAC is indeed very, very good. Most of us fail to consistently tell it apart from lossless in DBTs.

Proms program is impressively diverse. I am going to give it a listen! - Great tip - I downloaded the BBC SOUNDS app.

I'm not sure if you will get 320k aac outwith the UK. You may get 192k. However if you use a suitable VPN you may be able to cheat past that!

I'm finding the more 'sparse' sounds from the reduced orchestras particularly interesting as a different presentation of some music. The organ remains loud for the Poulenc, though. 8-]
 
It is a laughing matter that in order to defend the unneeded distortion by filtering and origami folding and unfolding of the signal that is introduced by MQA, Mr_Sukebe and others keep pointing to existing "distortion" or flaws in the replay chain that is common to all music files played through them.

you’re the person who’s asking for utterly bit perfect.
Just so I understand, does that mean that you don’t use any form of bass, treble, drc and use a NOS DAC?

just trying to remind you that the replay chain has many points where changes are made to the original source material.

in the case of MQA, it’s upfront, just as the “mastering” is.
After that, it’s a question of personal preferences. Don’t like it, don’t use it
 
The squarewave picture shows 75 fps. This corresponds to the compact disk time code with 75 frames per second.
The displayed time are to be interpreted as hours:minutes:seconds:frames. So e.g. 00:00:01:07.5 means 1 second 7.5 frames. 7.5 frames corresponds to 0.1s (=7.5/75). Result: 1.1 seconds.
The squarewave has a cycle time of 0.25 frames = 0.00333... seconds. This means we have a squarewave of frequency 300 Hz.

Now a squarewave consists of sinewaves of odd order = 1, 3, 5, 7, 9, 11, 13 ... With the given 300 Hz this results in 7 frequencies up to order 13 = 300 Hz, 900 Hz, 1500 Hz, 2100 Hz, 2700 Hz, 3300 Hz and 3900 Hz. We can now check the spectral lines and we can clearly count 7 lines up to 4 kHz.

This means: the displayed data are very correct.

Thanks for 'decoding' the timescales!
 
Is that before you insert it into an upsampling DAC which will then “invent” some data?

That hinges on understanding that the terms 'data' and 'information' have quite distinct meanings. Jim's Law, again.

Just as an analogue low-pass filter generates a given shape, defined from its input, so an 'upsampling' DAC can add new 'data points'. These need not create new 'information' though as a recording made in accord with the Sampling Theorem allows those points to detail the shape defined entirely by the input data to the upsampling.

Of course, the design/user may have chosen to do 'something else', in which case, yes, they chose to invent something different as the output. e.g. using a non-sinc filter shape for the reconstruction because they feel it 'sounds better'. Or use a NOS DAC with no output filter. Or use a Legato Link DAC.
 
Pleased to report that I've had a reply from 2L. And that they say the MQA versions of their files are created by feeding the DXD (as given on their site) directly to their MQA encoder. So we can assume any alterations we may find are due to the MQA encode-decode if we keep the later part of a system 'common mode'. That's helpful as it means we can use that to explore further at some point. :)
 
I'm finding the more 'sparse' sounds from the reduced orchestras particularly interesting as a different presentation of some music. The organ remains loud for the Poulenc, though. 8-]
It seems that the number of albert hall organs was kept at 1.
I agree that the "sparse orchestra" thing is interesting- it has been a feature of all the venues I have been to this summer. i particularly enjoyed the reduced forces at Glyndebourne for Katya Kabanova, and at ROH for Clemenza, slightly less so for Don Giovanni. I'm a little apprehensive about Tristan. Either way the "sparse audience" thing pre-freedom day was wonderful, if unsustainable.
 
My replay chain is a given thing. MQA is an unknown quantity - an extra filter process that changes the sound signature of the master file and changes it into a lossy file.
 
My replay chain is a given thing. MQA is an unknown quantity - an extra filter process that changes the sound signature of the master file and changes it into a lossy file.

By "given thing", that implies that you know what it's doing.
Do tell, I'm curious? You're asking for full deconstruction of what MQA is doing, so surely you'd want to know what your own system is doing?
 
My replay chain is a given thing. MQA is an unknown quantity - an extra filter process that changes the sound signature of the master file and changes it into a lossy file.

Let me know why this is a good thing.
 
My replay chain is a given thing. MQA is an unknown quantity - an extra filter process that changes the sound signature of the master file and changes it into a lossy file.

Let me know why this is a good thing.

I’m still going to ask my question, do you know what your OWN system is doing.
By not answering my question, I’m assuming not.

As for why you might want MQA to it, it’s the same reason that some DO use bass, treble, DRC, upsampling and a host of other processes that are applied between a microphone and your ears, because you “might” prefer it and “it might” better represent the sound in the studio/concert hall better.
 
Assume what you like. I have what you can see in my signature. It won't be changed. So it is given. I have the flavour that I like and plenty of purchased music stored in the formats I like. I see no need for a different flavour. If I want a change of flavour, I will control that myself and not let any lossy format choose for me. End of conversation unless you bring new insight to the topic of what exactly MQA is or exactly what it does to the files that are encoded. Ta.
 
I know what filters are used in my DAC or CD/SACD player, I can control upsampling through SOX for streaming if required through the Squeezelite commands used if I am playing ripped copies rather than physical media.
 
I’m still going to ask my question, do you know what your OWN system is doing.
By not answering my question, I’m assuming not.

As for why you might want MQA to it, it’s the same reason that some DO use bass, treble, DRC, upsampling and a host of other processes that are applied between a microphone and your ears, because you “might” prefer it and “it might” better represent the sound in the studio/concert hall better.
But people don't usually save their bass/treble/DRC-altered or upsampled files. They usually apply it on the fly. Instead they store the highest quality and least processed form of the files they can (I know I do).

That way, when better bass/treble/DRC/upsampling/special doo-dah processing becomes available next year, they can enjoy the improvement optimally.

You're comparing apples with oranges.
 
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But people don't usually save their bass/treble/DRC-altered or upsampled files do they? They usually apply it on the fly. Instead they store the highest quality and least processed form of the files they can obtain (I know I do).

That way, when better bass/treble/DRC/special doo-dah processing becomes available next year, they can enjoy the improvement optimally.

You're comparing apples with oranges.
I quite agree- or to look at it another way even the really hardcore folk at hydrogen audio think you should store music as flac because even transparent perceptual codecs cannot necessarily be transcoded transparently from one to the other.
 


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