First a question: are all equalizers in the digital domain the same in that they preserve phase by frequency, prevent additional jitter, and whatever else Ted works so hard to eliminate? I suspect not.
Can you design a fabulous piece of software that sits in my PC that I can manipulate… say invoke in Foobar or maybe catch the digital stream prior to player software, that uses your audiophile precision to give me a darn equalizer? A little panel pops with an OUT button, and then about four presents that I can click and just invoke or revoke? Maybe you know of an already software tool you can recommend?
I am sitting here drinking a few Pilsner Urquels listening to some old Monkees recordings through your Direct Stream Senior, and my ears are bleeding. I am dripping blood all over my white shirt. These recordings just hurt. They are so damn bright … they just suck.
Now I can turn on the equalizer plug in my foobar player and neutralize the evil, but I suspect the tool is not audiophile grade. Believe it or not, some of these Monkees recording actually have instrument definition and separation.
BTW, that guitarist on Valeri is amazing! (Louis Shelton, one of the Wrecking Crew!)
One of the evils of accurate playback is that… well it can be too accurate.
Bruce in Philly
If you stay in the digital domain then jitter isn’t an issue - the issue is simply how well your interface or DAC rejects that jitter. A more pressing issue is the noise (conducted and radiated) by the processor doing the processing - at times the more the processor is doing the more noise it makes…
If the material to be equalized is already in PCM then many (well implemented) EQs should be fine. If you want to EQ DSD then things can get more interesting: converting the DSD to lower rate PCM isn’t a good idea for a great audiophile EQ. Doing processing of full width PCM at DSD rates could easily be a problem for complicated processing. If CPU cycles are not an issue (perhaps by using graphics processors, etc.) you can use the same fundamental EQ algos… And a good sigma delta modulator for DSD to PCM isn’t rocket science these days.
It is the case that many filters used for various digital signal processing don’t take enough care with phase, amplitude, etc.
Is this a reason why SW designed filters inside an FPGA are (or can be if properly designed) superior in terms of phase accuracy?
The technology isn’t relevant except insofar as there’s enough horse power. Computers can do what I’m doing in the FPGA with careful coding and enough time. Some are fast enough for real time I suspect. Simply using floating point (or even double floating point) isn’t enough at places, but a good DSP coder knows what to do.
The problem comes when people try to save some work or time: some of the obvious changes aren’t as benign as DSP coding classes/textbooks might imply. And everybody has their own ideas about which filters to use in upsamplers and sigma delta modulators. I think mine are pretty good and I keep improving them usually by being more direct.
Was just searching for hardware equalisers and found this thread. I may start a new thread to see if anyone, at all, in this community actually uses equalisers (an evil, potentially, for HiFi, I’m well aware, but awfully useful, and sometimes a better option than not (see above OP!).
I’m very sensitive to tone, so I use an equalizer in Roon and Audrivana. Roon’s EQ is able to work with DSD, which is a plus, though I’m not sure how they do it. EQ’s are maligned in the audiophile community, but I just like them. I always liked hardware based tone knobs from the old days
Yes knobs preferable, digital increments ok if it has a decent display