MQA technical analysis

[bcwang]

It's graph time again. 2L-111 is a 352.8 kHz master with remarkably low high-frequency noise. Here's the spectrum of the original and the MQA reconstruction (decode + render):

 

[ATTACH=CONFIG]32642[/ATTACH]

 

According to my analysis of the renderer, this is the filter used by its 4x upsampling with this file:

 

[ATTACH=CONFIG]32643[/ATTACH]

 

It's a 24-tap minimum phase FIR filter. There is a notch centred at 88.2 kHz, which is clearly visible in the "rendered" music. Towards the high end of the spectrum, the filter again attenuates, and the shaped dither noise swamps whatever remains of the images there.

 

By render you mean the part of code that is adjusting the final signal to match the dac?

 

If this is just a digital analysis, one part that is missing is how the particular dac chip is handling an input that in digital form may look like this. Maybe the dac compensation process is to tailor the digital signal so the output on the analog side matches what came in on the analog side of this end to end "studio quality" process that is being claimed. To really know, an analog measurement of a MQA dac needs to be made to see resulting waveform rather than just what digital was being fed to it. Maybe at the end this particular dac distorts that waveform and it ends up looking like the pcm representation.

 

Did you post a digital representation of the post unfolding without the render portion? Maybe it is a closer match to the pcm equivalent.

[mansr]

By render you mean the part of code that is adjusting the final signal to match the dac?

 

I mean the part that does further upsampling after the initial decoding stage done by Tidal and supposedly soon other software. I have seen no evidence of any actual adjusting to a specific DAC, so I'm reluctant to describe it as such.

 

If this is just a digital analysis, one part that is missing is how the particular dac chip is handling an input that in digital form may look like this. Maybe the dac compensation process is to tailor the digital signal so the output on the analog side matches what came in on the analog side of this end to end "studio quality" process that is being claimed. To really know, an analog measurement of a MQA dac needs to be made to see resulting waveform rather than just what digital was being fed to it. Maybe at the end this particular dac distorts that waveform and it ends up looking like the pcm representation.

 

Did you post a digital representation of the post unfolding without the render portion? Maybe it is a closer match to the pcm equivalent.

 

The only way the rendering could be tailored to a DAC is by tweaking the upsampling filter coefficients. The effect of this would be to alter the frequency response. It would not "de-blur" anything. I checked the firmware for two different Bluesound products, and they both use the same filters. Then again, they are probably quite similar designs internally.

 

Here's a spectrum graph with the original, decoded, and rendered versions of 2L-111:

 

 

The decoded spectrum (red) obviously stops at 44.1 kHz. Below that, it has clearly higher levels of something, probably noise, than the original (blue). The rendering filter then pushes this down to the green trace. If you look at the frequency response of the upsampling filter above, it matches the difference between the red and green traces quite well.

[witchdoctor]

MQA-

Objectively with the songs I examined, the software decoder works well to reconstruct what looks like the equivalent 24/96 download.

 

Archimago's Musings: COMPARISON: TIDAL / MQA stream & high-resolution downloads; impressions & thoughts...

[Miska]

We don’t know exactly where is edge of accurate reproduction. Spectrum is infinite. So band should be infinite too, theoretically.

 

It is where the harmonics are not measurable anymore from the noise floor.

 

For example for the 2L recordings, harmonics reach to about 56 kHz, that's why I concluded that 60 kHz bandwidth is enough for those. But if you extend it to ~90 kHz of 192 kHz sampling rate or ~80 kHz of 176.4 kHz sampling rate you at least have enough margin.

 

As a result, having the filter fc above the highest harmonic means that the filter doesn't have time domain implications on the signal either...

[Miska]

MQA-

Objectively with the songs I examined, the software decoder works well to reconstruct what looks like the equivalent 24/96 download.

 

Archimago's Musings: COMPARISON: TIDAL / MQA stream & high-resolution downloads; impressions & thoughts...

 

Funny that the differences to original are higher with MQA and the file is bigger, than you would get with standard hires FLAC TPDF-dithered to 18-bit (-108 dB)... So completely no reason what so ever to use MQA, standard FLAC would give better quality and smaller file.

 

His tests gave good picture though that higher the amount of high frequency content, more there's difference to the original. But none of the tested samples really used frequencies above 22.05 kHz much. But there are tracks that actually have a lot of stuff there too. I've spent quite a bit of money to find such in MQA format.

[Miska]

By render you mean the part of code that is adjusting the final signal to match the dac?

 

If this is just a digital analysis, one part that is missing is how the particular dac chip is handling an input that in digital form may look like this. Maybe the dac compensation process is to tailor the digital signal so the output on the analog side matches what came in on the analog side of this end to end "studio quality" process that is being claimed. To really know, an analog measurement of a MQA dac needs to be made to see resulting waveform rather than just what digital was being fed to it. Maybe at the end this particular dac distorts that waveform and it ends up looking like the pcm representation.

 

Did you post a digital representation of the post unfolding without the render portion? Maybe it is a closer match to the pcm equivalent.

 

My recordings from analog output of a Meridian DAC with hardware decoding pretty closely match to what has been shown here for digital domain analysis...

[Miska]

Though very few speakers (headphones?) will reproduce most of that range. But it probably isn't a good solution to create problems further up the chain.

 

You can always extend the range using additional supertweeters.

 

There is another school of thought that says this is exactly the problem with hi-res: 44.1 captures all the audible effects of intermodulation with ultrasonics that occurred during the recording, so using higher rates is just asking for trouble.

 

I let those people have their RedBook. I still want my hires, preferably in DSD256 format. Thank you.

 

If extending range as little as to 100 kHz causes trouble, then people really should look for better gear instead of blaming it on too good source material. You can always have additional low-pass filter in your playback chain if you want to, but you cannot remove one that has been applied before.

 

If you put a filter close to 20 kHz or so, it will have sonic implications. Even 100 kHz is relatively close. 1 MHz is a good point. (I'm roughly in line with Spectral Audio on that)

[audiventory]

It is where the harmonics are not measurable anymore from the noise floor.

 

For example for the 2L recordings, harmonics reach to about 56 kHz, that's why I concluded that 60 kHz bandwidth is enough for those. But if you extend it to ~90 kHz of 192 kHz sampling rate or ~80 kHz of 176.4 kHz sampling rate you at least have enough margin.

 

As a result, having the filter fc above the highest harmonic means that the filter doesn't have time domain implications on the signal either...

 

I suspect, you refer to noise floor -120 dB (modern DAC).

 

About 20-30 years ago noise floor -120 dB was fantastic.

 

I can suppose, that 22 ... 25 kHz harmonics was deep into noise floor that time.

 

So 25 kHz could be considered as enought for reconstruction.

 

Let's look to probable future. There may be released DAC with noise floor -200 dB.

 

It allow to see (don’t hear) not only 56...90 kHz harmonics, but 150 kHz too. It is not real figures, of course, but suggested as example only.

 

In this case for reconstruction need filter with cut 150 kHz.

 

Also we can look further: when noise floor will -300 dB, as example.

 

What is edge there? When we must stop?

[mansr]

I found something interesting. The decoder has the provision to swap bits [4:11] and [12:19] in the output and signal to the renderer to reverse this shuffle. If this mode is triggered, only the top 4 bits are passed intact, so while the music would be recognisable, it would sound awful. I don't know what might enable this as I have not seen it with any of the samples I've tested (hardly surprising). Explain how this isn't DRM.

[mansr]

About 20-30 years ago noise floor -120 dB was fantastic.

 

I can suppose, that 22 ... 25 kHz harmonics was deep into noise floor that time.

 

So 25 kHz could be considered as enought for reconstruction.

 

Let's look to probable future. There may be released DAC with noise floor -200 dB.

 

It allow to see (don’t hear) not only 56...90 kHz harmonics, but 150 kHz too. It is not real figures, of course, but suggested as example only.

 

In this case for reconstruction need filter with cut 150 kHz.

 

Also we can look further: when noise floor will -300 dB, as example.

 

What is edge there? When we must stop?

 

Thermal noise puts the limit somewhere around -140 dB at room temperature.

[audiventory]

Thermal noise puts the limit somewhere around -140 dB at room temperature.

 

-140 dB open new details comparing current -120 dB.

 

But I made "Sci-Fi" assumption. There is opening of new details by lower noise floor matter.

[Miska]

I suspect, you refer to noise floor -120 dB (modern DAC).

 

I'm not talking about noise floor of DAC's, those are unimportant for the subject. What matters is noise floor of the recording, which includes acoustic background noise of the recording space, noise of the microphone preamps and the ADC. One can make recording at high enough sampling rate and then determine and decide sufficient distribution sampling rate at the mastering stage based on analysis of the final master. So what I was looking at is the original 2L's DXD master files.

 

It allow to see (don’t hear) not only 56...90 kHz harmonics, but 150 kHz too. It is not real figures, of course, but suggested as example only.

 

Yeah, 150 kHz is not a problem at all for DSD512 which can be pushed to have noise floor well below thermal noise for the 200 kHz bandwidth. DSD1024 is coming soon (and already used by some DACs).

 

So for the Sanken mic I was talking about, using the new ADCs running at 705.6/768 kHz PCM could be a good choice. Or why not DSD256.

[audiventory]

DSD1024 is coming soon (and already used by some DACs).

 

I'm agree. DSD1024 is nice thing for transferring 100 kHz band. May be more too, but I don't checked yet.

[Miska]

I'm agree. DSD1024 is nice thing for transferring 100 kHz band. May be more too, but I don't checked yet.

 

Well, it allows 400 kHz wide band with noise floor below thermal noise. For 100 kHz, DSD256 is enough.

[bcwang]

My recordings from analog output of a Meridian DAC with hardware decoding pretty closely match to what has been shown here for digital domain analysis...

 

Would you be able to post a pic of that? Just so it's easier to visualize. What about the flac version as well output from the same dac to see if it matches the theoretical? MQA vs theoretical analog measurement on the same chart.

 

By the way, what is the waveform measuring? Some noise level at a specific instant of the recording? It looks like near silence with the highest signal being -60db so it surprises me it doesn't match with the close correlation archimago's blog shows after decoding.

[Miska]

Would you be able to post a pic of that? Just so it's easier to visualize. What about the flac version as well output from the same dac to see if it matches the theoretical?

 

If you look earlier in this thread, I posted some results.

 

By the way, what is the waveform measuring? Some noise level at a specific instant of the recording? It looks like near silence with the highest signal being -60db so it surprises me it doesn't match with the close correlation archimago's blog shows after decoding.

 

Same noise is there throughout regardless of the position.

[mansr]

I'm still missing any "green-light" MQA files. If anyone has one, I'd appreciate a sample from it.

 

Has nobody seen a "green" MQA file?

[mansr]

@Miska, can you tell me if the Explorer2 activates the MQA rendering filter (as evidenced by high-frequency noise) when playing a software-decoded MQA file?

Does anyone have a definitive answer for this? If it can provide just the render function, I have some ideas for experiments.

[mm67]

Has nobody seen a "green" MQA file?

I think you might have a better chance of getting an answer in Bluesound Node2 thread

[mansr]

I think you might have a better chance of getting an answer in Bluesound Node2 thread

It's not specific to Bluesound.

[mm67]

It's not specific to Bluesound.

I see that know. If you read the comments at least one guy seems to have both blue and green led versions of one track : Computer Audiophile - My First 24 Hours With MQA

[bcwang]

If you look earlier in this thread, I posted some results.

 

 

 

Same noise is there throughout regardless of the position.

 

If you mean the spectrograms, I unfortunately still haven't figured out how to interpret them to draw any conclusions and especially not how to compare those with the graphs mansr has provided. Would you be able to explain?

 

What I was looking for was output looking like what mansr showed as digital analysis but of the actual analog measurement from an mqa dac. One playing the mqa file and one the high rate flac, to see that it did indeed change the output noise. I browsed the whole thread again and didn't see anything like that.

[bcwang]

I see that know. If you read the comments at least one guy seems to have both blue and green led versions of one track : Computer Audiophile - My First 24 Hours With MQA

 

That one guy being the founder of this site. =)

 

You know who to ask but I don't know if you'll get what you're looking for.

[Miska]

Does anyone have a definitive answer for this? If it can provide just the render function, I have some ideas for experiments.

 

I'll try to check this on Monday or Tuesday. I need to send the digital capture of the software decoded file because Tidal application enforces hardware decoding when it detects known MQA DAC.

[Miska]

If you mean the spectrograms, I unfortunately still haven't figured out how to interpret them to draw any conclusions and especially not how to compare those with the graphs mansr has provided. Would you be able to explain?

 

I thought it is apparent. Same information, but with one extra dimension - time. The Y-axis of mansr's graphics are color-coded and the frequency is swapped to the Y-axis because time is on X-axis.