Yes, I know but I thought you sold it and have now another NOS DAC...Holo May — I wrote a bit more about it on its own thread on here.
Yes, I know but I thought you sold it and have now another NOS DAC...Holo May — I wrote a bit more about it on its own thread on here.
I had a quick play with upsampling on my system yesterday. I changed the Audrivana output from PCM to DSD and then in the upsampling section specified DSD64.You should oversample/filter, unless you prefer the potentially euphonic effects of not using a reconstruction filter (ultimately it's your call).
The Spring performs best when fed DSD256, or you could try 20-bit 705.6/768kHz PCM.
I had a quick play with upsampling on my system yesterday. I changed the Audrivana output from PCM to DSD and then in the upsampling section specified DSD64.
As you may have noticed I don't really know what I'm doing, there are far too many options and things to setup for my tiny brain to figure out what they do or not. Anyway fumbling away with these settings, I found the sound to be very 'rolled off' at the top end and it was an instant dislike from my ears. If this is what DSD64 is supposed to sound like then I don't like it one bit.
So I went back to what I had before, PCM with settings like his:
Instantantly my ears preffered this. The top end is back as it should be there is no roll off and the sound is more open and the treble is back.
This simple try out of DSD64 has left with a distict impression that I don't want to go anywhere near DSD again. But like I keep stressing all this is beyond me and I don't have a clue what these parameters do, I just want something sensible.
If anybody can suggest something that may be an improvement on this which I may like even more I'm prone to give it another try and perhaps have an even more euphonic setup than I already have.
You'd have to know what you're doing, and I am one of those people who dreads having too much choice.
HQPlayer offers around 40 filters, half of them available in minimum phase and linear phase, and some 15 different ΔΣ modulators. More than enough in my view.
I'm now wondering about how easy it would be to write a prog that lets the user draw the required frequency-response and it generates the coefficients for sox to do the filtering...
Well, it's Winter time and the nights are long.
Perfect for "Etch-a-sketch" 8-]
Or are most people here too young to recall that?
All converter chips contain such filters.
Our investigation has revealed that these filters pose a threat to sound quality when they have insufficient ‘processing power’ to calculate the filter and its requantization at the required precision.
That looks like a really bad argument from a technical PoV. Using "insufficient 'processing power'" for a desired outcome is bad engineering. Plain and simple. You fix bad engineering by doing the engineering properly. By curing the disease. Not by applying some "sticking plaster" to alleviate its symptoms.Some comments from Grimm's engineers:
...
Our investigation has revealed that these filters pose a threat to sound quality when they have insufficient ‘processing power’ to calculate the filter and its requantization at the required precision.
For instance, the filtering is usually executed in several simplified, cascaded steps, which results in a loss of audio quality details.
Curous that an 'investigation' was needed to discover what is techincally called "the bleed'n obvious" *defined* by the wording of the assertion.
The 'solution' is also obvious. The system just needs to provide sufficient "processing power". Since the input information rate is finite, so is the required "processing power".
That looks like a really bad argument from a technical PoV. Using "insufficient 'processing power'" for a desired outcome is bad engineering. Plain and simple. You fix bad engineering by doing the engineering properly. By curing the disease. Not by applying some "sticking plaster" to alleviate its symptoms.
There's a lot of processing power available today (for which I would have kiiled in the late 1980s to early 1990s when I was doing some work in the field). And the engineering knowledge of how to do things properly in digital audio has been widely distributed via the professional journals. It's not something that's a closely guarded secret known only to a few equipment makers and gurus ... except when it comes to markering.
Why does anything outside the audio range matter?Unfortunately ASR's measurements are band limited to the audio range which makes them less capable of characterising performance.
Why does anything outside the audio range matter?
This is one thing that I've never quite understood when DACs are measured (or filters compared). Apart from the potential for high levels of ultrasonic energy to cause issues with a (badly designed) amplifier circuit, or blow tweeters, I don't see how it matters what happens outside the scope of what humans can hear.
Being caused by what though? Because that doesn't happen if you feed a speaker with such a resonant peak in the ultrasonic range with a normal analogue signal, (as evidenced by the literally tens of such speakers lab measurements in countless reviews over the years). Something else must be happening. So are you saying a DACs output has caused this to happen? or that it's happening in the digital domain somehow, because there's simply no mechansim in physics that I can think of that would cause it to happen in the analogue domain.One example of a tweeter oil-can resonance modulating into the audio range:
Well I must be deaf in that case (entirely possible), as I've never been able to hear aliasing. I've tried many times switching between CD/DAC filters that are supposed to have aliasing product and those that are supposed to remove, or significantly suppress them, and I hear no difference what so ever between them. At least not with real music.‘cause what happens higher up in the range you can’t hear gets reflected down into the audio band that you can hear.
I think it's cause by intermodulation but I am not the right person to ask.Being caused by what though? Because that doesn't happen if you feed a speaker with such a resonant peak in the ultrasonic range with a normal analogue signal, (as evidenced by the literally tens of such speakers lab measurements in countless reviews over the years). Something else must be happening. So are you saying a DACs output has caused this to happen? or that it's happening in the digital domain somehow, because there's simply no mechansim in physics that I can think of that would cause it to happen in the analogue domain.
Well I must be deaf in that case (entirely possible), as I've never been able to hear aliasing. I've tried many times switching between CD/DAC filters that are supposed to have aliasing product and those that are supposed to remove, or significantly suppress them, and I hear no difference what so ever between them. At least not with real music.
Has anybody actually ever done blind ABX testing to prove aliasing is audible?
One example of a tweeter oil-can resonance modulating into the audio range: