OBJECTIVELY, is there any sound improvement ??

[pkane2001]

 

2 hours ago, beerandmusic said:

I found this very good related article worth a read.

 

http://archimago.blogspot.com/2020/03/musings-which-measurements-matter.html

 

who suggests as i suspected that given amplification and room are in order that measurements of "modern day gear" are somewhat a non-issue...so i might as well just get something that checks my functionality boxes within my budget that sounds "good enough".   To my ears, besides amplifier and speakers, i rarely heard significant differences using ethernet, provided i was n the sweet spot and did room eq.

 

I know ASR is a dirty word around here, but there is a good thread there that tried to address something similar.

 

 

 

[pkane2001]

54 minutes ago, gmgraves said:

BTW, the latest Yggdrasil from Schiit measures extremely well, and sounds unbelievably good easily besting other contenders such as the Chord Hugo TT + the Chord M Scaler, or the Benchmark 3. 

 

In which measurements does the latest Yggy best these other DACs? Can you post or link to them? Just curious. I do like that Schiit started posting complete measurements for their products directly on their website. That's way better than most of the competition, kudos to them!

[pkane2001]

3 hours ago, PYP said:

Is the measurement of IMD a typical one for DACs and, if so, what is considered good performance?  Regarding IMD, is there any advantage to delta-sigma vs. R2R vs. FPGA-based (if that is the right terminology).  

 

I've read a bit about IMD, but don't fully get it.  Thanks.   

 

Harmonic Distortion and Intermodulation Distortion are very closely related. Both are caused by the non-linearity of the device. All analog devices have some non-linearity, so amps, DACs, preamps, headphones, and speakers all generate some level of HD and IMD. Digital devices can introduce non-linearities through computation, also.

 

Harmonic distortion is easy to see when testing with a single tone, measuring integer multiples of the fundamental frequency. IMD is easier to see and measure when two or more frequencies are present, since they interact to form additional frequencies by addition and subtraction, which in turn, interact, and produce more distortion frequencies.

 

Your DAC (Mola Mola Tambaqui) has very low HD/IMD levels, and so is very close to a perfectly linear device.

 

As far as I know, it's possible to produce a near-linear device using any of these methods, but the simplest one is S-D. R2R can also be made linear but with a lot more work (read: more money). 

[pkane2001]

2 hours ago, PYP said:

 

On your website, I didn't see that your Distort audibility tester provides IMD data.  Perhaps I missed that or it is the "more to come" part.  Just curious...  

 

DISTORT creates IMD by virtue of simulating a non-linear transfer function. With a single tone, this results in harmonic distortion. With two or more frequencies it would produce IMD. You can see this when you select a two-tone test signal.

 

I didn't put in a measurement for IMD, but certainly something that could be added.

[pkane2001]

58 minutes ago, pkane2001 said:

 

DISTORT creates IMD by virtue of simulating a non-linear transfer function. With a single tone, this results in harmonic distortion. With two or more frequencies it would produce IMD. You can see this when you select a two-tone test signal.

 

I didn't put in a measurement for IMD, but certainly something that could be added.

 

Here's an example. The same exact non-linearity tested with one and two tones:

 

One tone (1kHz) - harmonic distortion:

 

 

Two tones (18.5k/19.5k) - IMD:

[pkane2001]

1 hour ago, John Dyson said:

AIEEE -- the thing about digital filters and 'ringing' is all confused and silly.  Here are the absolute facts:

 

1)  The ripples that you see with brickwall FIR filters are NOT ringing, but they are Gibbs effect that results from a residue from MISSING frequency components.

2)The Gibbs type effect 'moves around' depending on the kind of filter (minimum phase, linear phase) based upon different delays vs. frequency, so the time location where the missing frequency components would have been.  So, the timing of the missing residue moves around relative to the rest of the signal.  (Kind of backwards wrt reality though.)

3) Even though some people say that you cannot hear the difference between minimum and linear phase, it MIGHT be possible in some circumstances -- but I haven't measured it myself...  Let me explain, a linear phase filter is 'perfect' except for a fixed delay vs. frequency.  A short (few taps) 'minimum phase' filter will have variable delay vs frequency, but a short filter cannot have much delay.   A very long (many taps) 'minimum phase' filter might have audible effects because of long relative time delays.  I don't know if they are audible, but a long (lots of taps) delay minimum phase filter MIGHT have audible artifacts.  It is up to those who measure these things WRT real world hearing tests to determine the answer'  @Archimago has  done some tests and I trust his results, whatever they are!!!

 

John

 

Filters can have some level of ripple in the pass-band (not ringing) That's determined by the filter design. For FIR filters it's often a tradeoff between the sharpness (slope) of the filter, amount of out-of-band attenuation, and the amount of pass-band ripple for the same number of taps. Here's a cool on-line resource for constructing and visualizing various filters: http://t-filter.engineerjs.com/

 

For example, 113 taps filter is needed to get the pass-band ripple to be under 5dB in this example:

 

And if I want to reduce the ripple to under 0.01dB, this requires 235 taps, all else being equal:

 

With modern, mega-tap filters, the ripple in the pass-band can be made fully insignificant.