Quote:
Slackman: I enjoy reading your posts. They are sort of a trainwreck, but of the fun variety!
Seriously though, I am all for you posting your "stream of consciousness" style impressions. Some people probably identify with your style much more than mine, so all types of reviews/impressions are good to have around. I don't know that I agree with every single one of your conclusions (specifically about ASRC not removing jitter....) but for the most part we are on the same page.
I'll say again that I don't think the ESS Sabre chip has a specific "sound" and that you might as well try any potentially good DAC that appeals to you. It doesn't matter if it sports a chip from ESS, TI, Cirrus, AD, or whatever. There are plenty of examples of good, bad, and mediocre designs using each of those chips so no reason to limit your choices.
What do you have your sights set on next? Any ideas?
Glad you enjoyed it
I can understand it makes for entertaining reading haha.
I was planning on doing it more clean this time, but the excitement of listening to a new DAC got to me it seems.
About the ASRC. If I understand correctly how it basically works..
The digital audio is basically "sampled" and interpolated to a new clock.
So all the errors (jitter/timing errors) in the original digital signal, are used "as is" to calculate the new sample points, which are then clocked with a new clock.
After this, only the jitter of the new clock is quoted. But the new signal is not bitperfect anymore and contains all the errors of the original signal.
This doesn't sound like real "jitter reduction" to me.
Now I'm not sure what other methods are used to actually reduce jitter before the interpolation.
(edit: the thing is, how would you know what the ideal perfect no jitter clock is of the incoming stream in order to relate it to the new clock? you don't know. you still have the same problem of underruns or overruns if you assume a certain perfect clock for the incoming signal. So it's still messy. Only now you're also adding on top of the jitter errors interpolation algorithm errors for the asynchronous sample rate conversion which are themselves audible like a bad upsampler. At least that's how I interpret the info I read and how I see real world logic of digital audio. If I'm wrong someone please tell.)
The white sheet of the Sabre DACs mentions a weird looking method of modifying samples close to the edge of sample changes or something.
Really useless info, can't make anything of it (perhaps there is more info on this somewhere else? or perhaps it is proprietary) but I don't trust it (it modifies bits). Though the Sabre DAC doesn't use "normal" ASRC since it used such a high sample rate to convert into.
That said..
Yes, I do think I know which DAC I'll try next.
The Anedio D2 of course
I'm very suspicious of it actually. No raving beforehand.
But I have hopes that it's errors in any areas are small enough to make it a DAC I can really work with well.
I'm ready to compromise. And I'm seeing and hearing all DACs have to compromise. And the D2 seems like a good bet to have been designed with sensible compromises, skilfully to a budget.
The only bad thing about it is that I'll have to pay the full price. No tax brakes for me when ordering from the USA (though import costs I should be able to get back)
It'll stretch the limits of my budget (if I'm guessing the price right). Will have to postpone certain room treatment and other things because of it.
