Audiolab 6000A and MQA file compatibility

Dawei88

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Hi
I’m using Bluesound node2i to stream music directly to the Audiolab 6000a connected by optical.
When I play a certified MQA 192khz album on bluesound or tidal app the Audiolab only show 96khz is being played.
Per my understanding, using optical cable on my bluesound is to let the amplifier DAC make the work and the Audiolab 6000a DAC is surely able to handle 192khz but for some reason it is limited to 96khz , how can I fully enjoy MQA file 192khz is there a setting to turn on somewhere ?
 

matthewpiano

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My understanding is that the DAC in the Audiolab (very capable though it is) is not MQA-certified and is unable to 'unfold' the MQA encoding. If you use an analogue connection between the Node and the amp, the DAC in the Node will do the unfolding and you will benefit from whatever additional quality the MQA format brings. Then it becomes a choice between which sounds better to you - non-MQA using the Audiolab DAC, and MQA using the Node's DAC.
 

manicm

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My understanding is that the DAC in the Audiolab (very capable though it is) is not MQA-certified and is unable to 'unfold' the MQA encoding. If you use an analogue connection between the Node and the amp, the DAC in the Node will do the unfolding and you will benefit from whatever additional quality the MQA format brings. Then it becomes a choice between which sounds better to you - non-MQA using the Audiolab DAC, and MQA using the Node's DAC.

Correct
 
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Dawei88

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My understanding is that the DAC in the Audiolab (very capable though it is) is not MQA-certified and is unable to 'unfold' the MQA encoding. If you use an analogue connection between the Node and the amp, the DAC in the Node will do the unfolding and you will benefit from whatever additional quality the MQA format brings. Then it becomes a choice between which sounds better to you - non-MQA using the Audiolab DAC, and MQA using the Node's DAC.
thanks a lot ! Very clear and i guess it solved my problem.
i just ordered a Chord company C-line RCA cable to connect the Node to the amp. Let see which DAC sounds better to me now.
 

Dawei88

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Ok so i tried both optical and RCA analog cable connected from the Node 2 to my Audiolab 6000A and i'm disappointed to not hear any difference.
When i using optical, the audiolab is showing me the audio quality (96kHz max even if the file is 192kHz ) so i expected to see 192kHz show on the amplifier when using RCA since it is the only way to let the DAC of the Node unfold it, but unfortunately there is no info displayed.
Do you know if it is normal that the audio quality is not displayed when using RCA cable?
I start to believe that the whole story about DAC quality and cable quality is pure illusion.
 

Deliriumbassist

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Ok so i tried both optical and RCA analog cable connected from the Node 2 to my Audiolab 6000A and i'm disappointed to not hear any difference.
When i using optical, the audiolab is showing me the audio quality (96kHz max even if the file is 192kHz ) so i expected to see 192kHz show on the amplifier when using RCA since it is the only way to let the DAC of the Node unfold it, but unfortunately there is no info displayed.
Do you know if it is normal that the audio quality is not displayed when using RCA cable?
I start to believe that the whole story about DAC quality and cable quality is pure illusion.

In regards to the Audiolab's display, this is very normal - in fact, it's expected. For every amplifier. RCA in this case carries an analogue signal, and analogue does not have sample rates and bit depths.

When you're connecting with optical, the Audiolab is doing the conversion, so the signal going in is digital, which does have sample rates and bit depths that is then converted to analogue inside the amp. It can see that information, so is therefore able to display it.
 
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Vincent Kars

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(96kHz max even if the file is 192kHz )
No, the maximum sample rate MQA can handle is 96 kHz even if the source has a higher bitrate.
When encoding a 24 bit / 192 kHz PCM file:
As the sample rate > 96 , down sample to 96
Downsample to 48, fold all audio frequencies above 24 kHz and store it below bit 17. (the audio origami part). As a consequence, all information there is lost.

On playback
If the media player is not MQA enabled, it will play as a 24 bit 48 kHz recording but effectively as a 17 bit recording because the folded part looks like silence.

If it is MQA enabled, it will unfold to 24 bit 96 kHz, restoring the frequencies above 24 kHz.
Effectively is will play as a 17 bits file because the part below 17 has been used for the origami part.

The last "unfolding" is only allowed in hardware (the DAC is MQA certified because a license has been bought)
In this case the DAC recognize the watermark.
It will apply the MQA prescribed minimum phase filter and as the original file was 192, applies oversampling as well.
So now the display says 24 bit /192 kHz but effectively you are listening to a 17 bits /96 kHz recording that is upsampled.
 
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Dawei88

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Yes, there is no difference to my ears between the Node 2i DAC and the Audiophile 6000A DAC. I will be using my optical cable as i like to see the the amp working and display the bitrate.
When using the RCA cable and so the DAC of the Node 2i i have no idea if the DAC is actually doing any proper job to send the signal in 192kHz and since i cant hear any difference it is just money wasted in this RCA Chord cable as i already had the optical cable before.
I really though there would be a advantage of using good RCA cable over optical but i guess i had wrong or illusionist expectation.
 

Wil

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Say it's-a-time-effort, Premier League is equivalent to Hi-Res Audio. Will the addition of a player always significantly change a club's performance? May be the-club excels only in the First Division.

And consider this:
"What we need are audio components that are capable of delivering High-Resolution Audio. For example, the D/A converter (DAC) built into the typical DVD player usually has a "24-bit" DAC that only delivers a 90 to 95 dB SNR. This is the equivalent of 15 to 16 bits (CD performance at best).

But, an outboard DAC is only a partial solution to the High-Resolution Audio dilemma. A second key part of the problem is the performance of the audio power amplifier. A 24-bit audio system is useless if it passes through the typical power amplifier. It is nearly impossible to find power amplifiers that can deliver an SNR higher than about 102 dB. This is the equivalent of 17 bits (adequate for CD applications, but definitely not adequate for High-Resolution Audio). Anyone who thinks they can hear the difference between 16-bit and 24-bit digital audio through a "17-bit" power amplifier is fooling themselves."

And yesterday, I was reminded that WHF's current hi-fi reference system [has] a £36,150 pre/power:
 

Dawei88

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And consider this:
It is nearly impossible to find power amplifiers that can deliver an SNR higher than about 102 dB. This is the equivalent of 17 bits (adequate for CD applications, but definitely not adequate for High-Resolution Audio). Anyone who thinks they can hear the difference between 16-bit and 24-bit digital audio through a "17-bit" power amplifier is fooling themselves."

From what i can see the Audiolab 6000A DAC is capable or are we talking of something completely different ?
Max. Sampling Frequency192kHz
Signal-to-noise ratio (S/N)> 112dB (A-weighted)
 

Wil

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I neither agree nor disagree with that opinion from John Siau, it’s just a viewpoint to explore. What you would’ve read if you had linked through:
In contrast, the Benchmark DAC2 outboard DAC has an SNR of 126 dB (equivalent to 21 bits). The DAC2 is actually capable of delivering peaks that are 3.5 dB higher, giving the DAC2 a total SNR of 129.5 dB (21.5 bits)...

For this reason, Benchmark has undertaken the difficult task of building a power amplifier that is worthy of the title "High-Resolution". The new Benchmark AHB2 power amplifier has a 130 dB SN.


As I said earlier:
Hi-Res Audio, simply, keep trying, or give up.


I won’t have time to reply for the next 10 hours.
 

Dawei88

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I neither agree nor disagree with that opinion from John Siau, it’s just a viewpoint to explore. What you would’ve read if you had linked through:
In contrast, the Benchmark DAC2 outboard DAC has an SNR of 126 dB (equivalent to 21 bits). The DAC2 is actually capable of delivering peaks that are 3.5 dB higher, giving the DAC2 a total SNR of 129.5 dB (21.5 bits)...

For this reason, Benchmark has undertaken the difficult task of building a power amplifier that is worthy of the title "High-Resolution". The new Benchmark AHB2 power amplifier has a 130 dB SN.


As I said earlier:



I won’t have time to reply for the next 10 hours.

not sure how I should interpret your messages but this time it seems you are not really here to help me ,so I will follow your advice and give up on you.
 

Wil

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not sure how I should interpret your messages but this time it seems you are not really here to help me ,so I will follow your advice and give up on you.
Thanks, I'm content that you don't comprehend me, and that I'm freed.
"Happiness may well consist primarily of an attitude toward time."

I understand Cantonese, there are sufficient CantoPop SACDs to amuse me for years to come e.g. my last purchase:
7-3-sacd-jpg.2205

I've evaluated their Hybrid CD layer on my many disc players (including CD-7)…

Music-listening, I simply will devote my funds (and time) towards (exploring) Hi-Res recordings.

Name some 16bit/44.1 only recordings that may tempt me to own.
 
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D

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not sure how I should interpret your messages but this time it seems you are not really here to help me ,so I will follow your advice and give up on you.


What I think Will is trying to say is, stop looking at the numbers and use what works for you. The differences are subtle, always have been and you have to have speakers that are able to render those differences ie are your speakers "full range" speakers and even then it's not an easy thing to hear. But make no mistake the difference is there, just not like some describe/are so poetic about.

At best hi res audio sounds a little fuller, has little more flesh on the bones. MQA with 2 unfolds in software will work through your DAC with no problem. Would I have bought a 4-500 quid streamer for it? Nope.

What will really scramble your eggs is when you realise you could have achieved the same performance or arguably better with a 70 quid piece of software from Audioviranna for your computer enabling 2 unfolds in software. But you pay for connivance we always have done. Not many want a laptop hanging out there DAC all the time.

The bottom line, have fun and listen through the streaming service of your choice and don't sweet the numbers.
 
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Wil

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Thanks millennia_one.

Let's review my input into this Thread, I simply referenced a WHF insight on "Optical is usually restricted to 96kHz." That's actually all I wanted to say because I anticipated performance differences won't likely be perceptible.

Then Gray said:
Don't worry, you're not alone.
Audible differences between DACs can be very subtle, between sample rates, non-existent.....and audible differences between cables?....let's not talk about that :rolleyes:
And I replied


There's a 5-star WHF review of the 6000A, saying many flattering things:


To be less oblique, I believe Hi-Res Audio, to fulfil its potential, needs combining of a (much) higher price point system/setup. What he has can't rise above CD-quality-results.

Take a parallel analogy, I still have a PD5050D Plasma monitor, it plays 1080p. But converted to 1080i into its 1365x768 resolution-limit.
 
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Deliriumbassist

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Yes, there is no difference to my ears between the Node 2i DAC and the Audiophile 6000A DAC. I will be using my optical cable as i like to see the the amp working and display the bitrate.
When using the RCA cable and so the DAC of the Node 2i i have no idea if the DAC is actually doing any proper job to send the signal in 192kHz and since i cant hear any difference it is just money wasted in this RCA Chord cable as i already had the optical cable before.
I really though there would be a advantage of using good RCA cable over optical but i guess i had wrong or illusionist expectation.

If you can't hear the difference, then the Bluesound is doing the business. DAC quality aside, what you're changing is where the conversion happens, that's all. Analogue signals are not described in bit depth and sample rate because they are continuous signals - think watching a flowing river with your own eyes, constantly changing. Digital is storage, and is made up of snapshots of that river, which an amplifier (as in the actual amplifier stages that modulate power) cannot use. The Audiolab DAC is doing the exact same thing - digital goes in, analogue comes out.

As you can't hear a difference, I would probably use the Bluesound DAC because the Node 2i is likely able to handle more filetypes than the Audiolab (basing that on the fact that the Node 2i can handle MQA, and that Audiolab does not specify what file types it can accept in the manual). Plus if the Node is your only source, you can set the Audiolab to Pre-power mode, which basically disables the Audiolab's preamp so all control, including volume, is done strictly from the Bluesound app. I agree with what's been said - try not to let the numbers and/or lights flashing up on equipment affect your enjoyment - the specs of the equipment tells us that both DACs handle 24/192kHz just fine.
 
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Dawei88

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Thanks millennia_one & Deliriumbassist
it is clearer for me now.
you are right, i shouldn't let the numbers affect my enjoyment. Maybe i can explain a bit more why i focused on those numbers, as i'm using KEF Q150 with my audiolab and node 2i, overall i'm happy with the sound but the bass is a little bit trouble me and i thought i would obtain a better performance (perhaps even better control) if the source is being played at fullest bitrate. I understand now that it is not that simple and probably only future upgrade to better speakers will give me the full satisfaction.
I will follow your advice Deliriumbassist , as my only source is the Node i will use the bluesound DAC so use my analog cable.
 

Deliriumbassist

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Thanks millennia_one & Deliriumbassist
it is clearer for me now.
you are right, i shouldn't let the numbers affect my enjoyment. Maybe i can explain a bit more why i focused on those numbers, as i'm using KEF Q150 with my audiolab and node 2i, overall i'm happy with the sound but the bass is a little bit trouble me and i thought i would obtain a better performance (perhaps even better control) if the source is being played at fullest bitrate. I understand now that it is not that simple and probably only future upgrade to better speakers will give me the full satisfaction.
I will follow your advice Deliriumbassist , as my only source is the Node i will use the bluesound DAC so use my analog cable.

By trouble do you mean too much? Boomy bass is much more to do with the room and the relative position of the speakers to the wall. Are they close to the wall or in the corner? Pretty sure the KEFs come with foam plugs for the bass port - they certainly did many moons ago when I was a retailer for them. Bring them off the wall to help reduce the roominess, or if you can't bring them off, stick the foam in the ports, and that should help alleviate.
 
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As mentioned above if the bass is a little strong move the speakers around a little or bring your listening position forward a little. THOUGH I've never had luck bunging the ports it robs the speaker's of way to much especially if they aren't a 2 stage bung. Maybe as an experiment throw a few cushions in the corners of the room and behind the speakers to see if they tame that bass. if they do maybe look into so bass traps ie bookcases, more soft furniture and so on.

I would say if you want some advice on speaker placement is perhaps start another thread with that question in mind.

But in answer to your question is, no, a source won't help with this sort of thing, think of it as a spice to add some flavour, in fact it could make things worst.

a new amp might help to offer more control of the speakers. but try the free things first.
 

Wil

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I would say if you want some advice on speaker placement is perhaps start another thread with that question in mind.
Whatever OP's thoughts on me, recalling WHF for the reading community:

And I once quoted MS:
In honour of Ken's friendship with Malcolm Steward (who has passed away too), I'll reproduce MS's public-piece which is lost due to his Top Audio Gear site being [unfunded]:
Opinion: Severe Loudspeaker Alignment (February 20, 2014)
I have just had two troublesome loudspeakers in for review. Each of them, when installed in a normal fashion in my music room, exhibited a tonal balance – or, more accurately, a tonal imbalance – that suggested that they had been voiced for the American market. A UK loudspeaker designer once warned me off listening to a couple of models in a range he had designed telling me that I would not appreciate them because they had been “voiced for the Yanks”. Strangely, he subsequently wound up working for JBL, one of America’s most successful and prestigious loudspeaker brands.

That designer’s “Yank” voicing involved cranking the frequency extremes to give the loudspeaker bass that invited involuntary bowel movements in the listener and treble so strident that it could strip the enamel from one’s teeth. The balance was way too brash and lacked the sophistication I seek in a loudspeaker – and I have a forgiving nature where most loudspeakers are concerned, and I can even listen past the colorations of most horn designs.

Regardless, writing reviews for print magazines these days tends to be an exercise in finding the positive aspects of any product. The reason most publishers give for this is “if it is not worthy of our readers’ time, why are we bothering to write about it and squander pages on it?” My countering of that opinion could take up many pages and would only be of interest to similarly disgruntled wordsmiths so I will refrain from biting the hands that feed me – albeit only after a fashion. Times – and editorial policies – have changed dramatically since the days when I first edited Hi-Fi Review magazine.

Getting back on topic, the ‘normal’ set up for loudspeakers in my room involves placing the enclosures a couple of feet in front of the wall I face when listening and about the same distance from the side walls. I usually toe in the speakers so that their axis cross just in front of my seated listening position. I can not be bothered with acoustic room treatments beyond carpeting the floor and always having the curtains behind my listening seat drawn closed. I have no wish to have bass traps/absorbers in the room corners nor anything equally as unsightly anywhere else, thank you. Room treatments are for studios, where they are not concerned with enjoying the music. I have been in control rooms in studios such as Abbey Road, where the sound sucks donkeys through a straw. It might work for the likes of producer, Alan Parsons who played me his then latest album there, but it did nothing for me and, I believe, I told him so. If I remember accurately, this did not amuse him. Face it, studio engineers generally do not know a lot about getting the best from their equipment. I have seen tape decks running off extension leads purchased at the local B&Q, and B&W monitors being auditioned with their bass units still masked by the flight cases used to transport them because the tech was too lazy to pull the speaker completely out of the case because he thought that “it made no real difference to the sound.

Begin by placing the loudspeakers in the ‘normal’ position then address boundary reinforcement considerations, remembering, of course, that both the wall behind the speaker and the one to its side exert an influence on its bass output. If there is too much low frequency information edge the speaker away from those walls, perhaps by as little as an inch at a time. When you are dealing with LF it is always best to have the speaker properly spiked into the floor, of course, even if it can make manoeuvring the enclosure tricky for you. Just be careful that you do not lower that spiked, weighty cabinet onto your feet.

Then it is time to tame that treble with toe-in. Twist the enclosures so that the tweeter axis crosses at a point in front of your face when seated. The dispersion characteristics should ensure that your ears are not starved of treble information. The treble should not, however, make your ears bleed because you should now be out of the ‘direct line of fire’ of both units and just be hearing their peripheral output. This amount of toe-in might look odd or severe to anyone who has not been to one of Ken Ishiwata’s demonstrations. I say this because Ken, a world-renowned listener – is an adherent of quite severe, dramatic-looking toe-in when he sets up loudspeakers. It often helps if you can sit and listen while a companion adjusts the loudspeaker positioning.
As well as ameliorating tonal fierceness issues with some loudspeakers the arrangement described can also help focus the stereo soundstage and deliver not only improved positioning of instruments but also better focus and more solid, secure images.
 

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