Sound Quality Vs. Measurements

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Depends on the desired transfer function and what the circuit bits comprise. I don't think there's a single one-size-fits-all answer, which is why both exist.

I should have been a bit more specific: I was referring to phono RIAA eq/amps, and specifically to the only.

My question was prompted by the very fact that both types live and thrive, which obviously means they both have their proponents and detractors.

A question of definition: if I simulate an inductor with a gyrator, then capacitively couple to it to make essentially an LC filter circuit equivalent, is that active or passive?

A goor dquestion, really makes one wonder. Personally, I'd call it active, but that's just me.

Actually, I'm considering this point because I should be getting around to it in a few months' time. I don't want to rush into it, and I appreciate the fact that collective experince has much merit. Some of us have been at it for decades, as opposed to me, I just made a few according to other people's designs, and about the most I added was improved PSU filtering. No big deal.

That's why I'd like to hear people's opinions, so I can make up my mind as to what exactly do I want, then do some research on what I want and finally do it, but do it right. I would like to get it right the first time, but I'm aware of the fact that "doing right" is also relative to one's tastes. That's a risk I'll have to take.
 
About DAC, have you given up the idea of non-oversampling?

No matter what, NOBODY should give up the idea of "real time" DACs.

I bought one from an Ausiie DIY company at what I'd call a very fair price. It uses 8 Philips 1543 DACs in parallel, and an AD 847 output op amp. Highy oversized PSUs, one for the digital and another for the analog section. Possible inputs S/PDIF or light.

It has a sound which does not remind one of digital playback at all. To use a much misused phrase, it sounds completely analog. The only thing which makes you pasue and think is the fact that there are no clicks and pops typical of LPs.

Over the years, I've tried it with many a CD player, with expectedly different results. The cheaper the model is, the greater the difference, for the better of course. Sometimes, it's there but is not impressive, other times it socks you in the eye.

I'll look up their address.
 
DIY Fidelity

That's it, but something seems to be odd with their site?

In a recent interview, he stated that the issue was not whether it is Non-Oversampling or Oversampling, but the use of the digital filter. Digital filters normally cut off the signals beyond 20kHz with a very steep curve and need around 2 milli seconds of time to calculate the enormous data. This is where it causes smearing in the time domain.
:Pinoc:
 
I bought two pairs of Dyna A25s in the 1970s and still use them for video and background music.
Seas is now selling an A26 kit, though it's more much expensive than the $66/pair original cost.

https://www.madisoundspeakerstore.c...6-10-2-way-kit-pair-based-on-the-classic-a25/

I bought a pair recently to renew my experience of 40 years ago. I was bamboozled into selling my pair. After a brief period of disappointment I fell back in love with them. They sometimes totally win me. When they don't I am still astonished how little has really improved when modern designs. Only The Celestion SL6 is a better design to my ears. Incredible as it is one of the few metal tweeter designs I like. Celestion had trouble and made sure it worked ( 22 kHz notch filter ), the metal box one is not my cup of tea. The red Rose Rose Rose Bud if I ignore price also, alas one can't ignore it.
 
I would like to suggest that there is a sort of formula that optimizes tonearm-cartridge resonance. It is 10Hz with a Q of 2 (6db boost at 10Hz). Of course, this is only a target, but it works very well for a number of reasons.

That seems about right.

Reading the AES papers it seems 20 Hz - 15 dB and 100 Hz - 10 db is the natural performance of the lathe. The EQ used will allow more bass or more cutting time. If the stamper is burnished the response is - 3dB 10 kHz. All of this has to be allowed for. There is then cutting head resonance which should be audible and thankfully usually is well hidden. Theses same problems seem like ceramic cartridges. The cutting late head like the ceramic cartridge will be very stiff when ideally it would be best not to be. In the case of the lathe that is just how it must be.


If one adds all the cutting problems together it should be a non started for hi fi. Even in 1927 despite the known problems the recordings have a bit of magic. Blumlein was given the job of busting the RCA patent for cutting lathes. Not content with cloning the process he improved the rigidity resonance compromise. The Blumlein cuts are some of the least EQed examples of all. When his boss Issac Schoenberg lent him to 405 line TV and radar he said the 1/2 penny royalty saved alone meant Alan never need do a stroke of work again to pay him back. Alas radar led to his death. Alan's system was quoted by Goering as shorten the war by 2 years I have been told. It was the less sophisticated due to his death than competitor systems, but was up and running. Churchill called it H2S. My dad taught me radar when I was 4 and the electronic associated with it. I can still activate him on it if I try hard enough. He was the last in the UK who could operate CH. Later on he found out what he taught me was classified and he had signed the official secrets act!

This is why CD disappoints me so much. With so many problems LP is better. No way should that be true.
 
What is smearing in the time domain? And can it happen without smearing in the frequency domain, or are the two related?

I have heard that term before, but what is it? Its sounds dirty, but not in a good way.


Not sure if this helps? The problem as with all null tests is insuring no null error. This type makes it more complex. To me what you say seems logical. As someone said. Crudely you turn the graph by 90 degrees and that's all. I doubt it is but seems reasonable.


http://www.nordost.com/downloads/NewApproachesToAudioMeasurement.pdf
 
diyAudio Senior Member
Joined 2002
Hi,

Very interesting read but doing the math using the figures on page #9:

Quotes from the Nordost .pdf file:

In fact, that reduction in misplaced samples amounts to 36%. Repeating
the process with the Kinibalu support reduces the error by a further 15%,
although bear in mind that this result is cumulative – the Kinibalu had
less to work on because the power cord had already improved things

And:

Finally, we switched on the Quantum unit and once again, we were able to
measure a further decrease in error – this time, another 11%, resulting in an
overall reduction in timing error with all three upgrades in place, of 52%.


36 + 15 + 11 = 62 and not 52 as stated.

Cheers, ;)
 
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diyAudio Senior Member
Joined 2002
Hi,

That's why I'd like to hear people's opinions, so I can make up my mind as to what exactly do I want, then do some research on what I want and finally do it, but do it right.

No idea whether or not it is going to be helpful but in valved electronics passive RIAA EQ is increasingly popular lately.
Either a single EQ block sandwiched between two gain stages or, even more popular is to split the block in two and doing the EQ over two gain stages.
The most obvious reasons I can think of are simplicity of calculus and reduced loading of the stages.
There are probably other pros (and cons) to it.

Typical active correction employed (usually negative) feedback around the gain block using Cs and Rs in the loop to achieve the EQ.
This is far more complex and hence less popular nowadays.
One of the main concerns while using valves is that their parameters can (and do) change as they age making the EQ less accurate as time goes by.

The of course you can combine both methods and all kinds of trickery, the list is endless...

No idea how this all works in the sand world but I suppose passive RIAA is likely to be popular as well?

Cheers, ;)
 
Tattoo, Nigel, thanks, but the point I was making in my own vague and indirect way is: "smearing in the time domain" is vague audioramble without any technical meaning.

The addition of the term 'time domain' does give the impression of technological prowess, but in a completely nonsensical way. Because the same phenomenon would have an impact on the frequency domain as well. So, my advise to those prone to marketingspeak is the following. Henceforth, speak of "smearing" or "smearing in the time ánd frequency domain", but please realize when you do so, it still means nothing at all.
 
Tattoo, Nigel, thanks, but the point I was making in my own vague and indirect way is: "smearing in the time domain" is vague audioramble without any technical meaning.

The addition of the term 'time domain' does give the impression of technological prowess, but in a completely nonsensical way. Because the same phenomenon would have an impact on the frequency domain as well. So, my advise to those prone to marketingspeak is the following. Henceforth, speak of "smearing" or "smearing in the time ánd frequency domain", but please realize when you do so, it still means nothing at all.
:cheers:
That site uses "sciency" speak to make potential customers think they buy a superior product. When in fact they buy a product that outputs large amounts of high frequency noise at very high levels, very likely resulting in large amounts of IM distortion.
But of cause if someone likes that, I'm fine with it.
 
:cheers:
That site uses "sciency" speak to make potential customers think they buy a superior product.

If you are referring to the site posted by dvv, I think the site didn't do "sciency" speak nor lies. They are neutral by saying what has been said by the person who brought the idea of non-oversampling with paralleled TDA1453.

That they said "In a recent interview, he stated that the issue was not whether it is Non-Oversampling or Oversampling, but the use of the digital filter" is not an effort to sell anything.

Ryohei Kusunoki indeed said that the oversampling requires digital filter and often the digital filter is of insufficient quality so to create "jitter".

So do you think Kusunoki was wrong? Or the site purposely misinterpreted Kusunoki??

When in fact they buy a product that outputs large amounts of high frequency noise at very high levels, very likely resulting in large amounts of IM distortion.
But of cause if someone likes that, I'm fine with it.

Agree that a few Hz above 20kz can be audible/sensible but how to proof that it will produce sufficiently large IMD (any reference?)?
 
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