MQA technical analysis

[Miska]

I doubt any of the select few with access to encoders would be willing to do this. If they did and MQA found out, they might find said access suddenly revoked. Is it even allowed to use an encoder unsupervised by MQA?

 

They won't gain large scale user base if they would really want to know about ADC and mastering, unless they provide widely available plugins for DAW's like ProTools and Pyramix.

 

If person X publishes measurement results and doesn't tell who encoded the data, they won't have any means figuring out who did the encoding unless they store and inspect everything that was ever encoded to MQA. And if they do that, if I were recordig house, I would stay far away from MQA. But based on their FAQ one can make internal encoded variant for listening that doesn't go through their immense signing/DRM hoopla. And that's enough.

 

I really don't understand who could have slightest interest on some MQA big brother watching over and dictating all their stuff. Although all the Orwellian stuff seems to be in fashion these days I just feel sick of such.

[Miska]

It has been a while since I read the patent and white paper. Was there something about a windowing function (doesn't mean they're currently employing it)?

 

Triangle ... how crappy is that!

[Miska]

I couldn't hear squat above 16KHz a couple of years ago and it's probably lower now. I'm assuming aliases will cause no audible harm to anything I'm listening to unless those above 16KHz cause harmonic or intermodulation distortion at audible levels in lower frequencies.

 

Any indication MQA might do harm in this way? Or in some other evidently audible fashion?

 

It is system dependent, so there's no single answer...

[Miska]

The certification is required because MQA have to brew up the renderer filters that undo a particular DAC's sins.

 

Have you seen any evidence of such? At least the Explorer2 vs Brooklyn look the same. Luckily Brooklyn has proper filters for non-MQA listening too, plus DSD support.

[Miska]

46 minutes ago, Shadders said:

I looked at the Pacifics Microsonic offerings, and no mention of the ADC filter.

 

PMD didn't have digital filter for 176.4 kHz ADC rate. But when output was set to 44.1 kHz, it uses normal FIR anti-alias filter for the rate conversion. I don't remember the FIR in more detail...

 

[Miska]

2 minutes ago, Shadders said:

Hi Miska,

Thanks - i was referring to the analogue filter on the front end of the ADC - do you know anything about that ?

Thanks and regards,

Shadders.

 

I don't really remember much... It is likely not delta-sigma ADC, so it would need some more aa-filtering in analog domain. But of course it depends how much designer expect there to be content up to Nyquist frequency and how much they are ready to sacrifice bandwidth for the filter transition band. I think PMD was optimized for CD pass-band, so I think it started rolling off somewhere near above CD's Nyquist.

 

MQA seems to be working in a similar way, by sacrificing band above 22.05/24 kHz for the filter roll-off to keep the filter very short. So instead of more ringing it makes the rise-time longer. So while winning transient domain from ringing perspective it loses on rise time.

 

Normal studio gear uses typically second order low-pass with -3 dB point around 100 kHz, since they use delta-sigma converters and there's no need to do a lot of analog pre-filtering since most will be done in digital domain. Then you may have bunch of options for the digital filter.

 

[Miska]

22 minutes ago, Shadders said:

I was thinking of the 1980's and 1990's before sigma-delta - when the album was recorded direct to digital, or analogue tapes transferred during this time.

For the Tascam - is there in information on the filters (i assume sigma-delta, then filtered) ? Thanks.

 

90's was already SDM converters. Chips from same vendors as these days, TI, AKM and Cirrus Logic.

 

For pro-audio gear, AKM and Cirrus Logic seem to be most popular.

 

I still have some tens of these older DSD-capable true 1-bit ADC chips on tape, originally from 2001:

http://www.ti.com/product/pcm1804

 

[Miska]

1 minute ago, Shadders said:

Thanks. I checked - late 90's seems to be the earliest - but in any case - i can still progress with assumed FIR filters.

Do you know how many taps they will have had? (256. 512 ???)

 

Those are cascade FIR designs, just like DAC chips. Last stage to 44.1 kHz less than 200 taps, preceding stage usually 25-50 taps (to 88.2/96k) and the stage before that less than 25 taps (to 176.4/192k).

 

[Miska]

1 hour ago, Fokus said:

As others said, from the early-mid 90s on all sigma-delta stuff, with first or second order analogue pre-filtering and then digital half-band anti-aliasing filtering in the ADC's downsamplers.

 

And before that linear ADCs sampling at baseband, and thus with analogue AA filters, something like 8th order elliptic.

 

The SDM ADCs with their HB filters are easy to recognise: do a spectral analysis of a CD, when the spectrum runs right up to Fs/2 you are looking at HB: there is always a bit of aliasing.

 

The older ADC class is typefied by the Sony PCM1610/1630 systems. I have quite a few older CDs where the spectrum reaches a lot of attenuation at Fs/2, i.e. free of aliasing. Remarkable.

There is not much data on these old ADCs available on the net, but here is some

Sony DAT review.

 

Yeah, although with recent material the recording is quite typically done at 96 kHz sampling rate and then mastered for CD with software conversion from 96k to 44.1k. That's what also brought us the comparison site: http://src.infinitewave.ca

 

I have good example of this too...

 

Here's early (first?) CD version of Pink Floyd's DSOTM, "Time":

 

Here's 30th Anniversary SACD's RedBook-layer:

 

And here's the latest remaster:

 

And for comparison, 30th Anniversary SACD's DSD-layer:

 

 

1 hour ago, Fokus said:

If I remember correctly the Pacific Microsonics ADC (or rather, the entire system including HDCD encoding) used 2x oversampling, so part of the CD-production AA filter was done in the digital domain.

 

It runs at 176.4 kHz, so 4x oversampling.