Out of curiosity have you spent some time listening to some NOS DACs? Reading back through the comments, seeing the graphs and the case being closed for you, so wasn’t sure. Which was my point, why does it seem to be preferred by some, that’s the science explanation I’m curious about.
2 Likes
Is that a hill you want to die on?
So you are saying humans cant hear spike pulses (like gun shots) either because to represent them, you need an infinite number of sine waves?
Just wanted to say using minumum phase is more modern, and sounds better.
Minimum phase puts the whole artifacts after the impulse.
So when the bass kicks, it kicks as it should and the echo is after the kick (just like real life). Linear phase has some echo before the signal which is weird and unnatural.
Linear-phase is only suitable for very limited occasions where you have two signals of the same things and want to change them separately. Like when you want to EQ the left channel and right channel differently. That’s where you need the phase to be linear.
That’s why iFi nano has minimum phase for “listen” option. Because that’s how things sound in real life. First the sound, then the echo.
Yes. More specifically, “instant” pulses that rise and fall extremely fast are not heard as such because that would require ultrasonic hearing. We hear them as slower rises and falls, as gentler slopes at least 50 microseconds in duration, because that’s all our 20 kHz hearing allows. It’s useless to try to reproduce them as 5 microsecond transients because they will still be heard as 50 microsecond transients anyway (in fact this is how some NOS fans argue that nothing bad will happen because of the lack of a reconstruction filter: human hearing has its own low-pass filter at 20 kHz that will be applied to all incoming signals anyway).
Haven’t seen any evidence that pre- or post-ringing can be heard, so I don’t see the relevance if there’s some of it before the pulse. But again I’ve seen what the minimum-phase filter does to the shape of the signal in the time domain (especially the higher frequencies) and I didn’t like it:
I don’t think science has humans modelled and explained at such a level as to say why people prefer things. It involves too many things, from biology to neurology to life experiences to personality types etc. You’d have to get together a multi-disciplinary team from many scientific fields to get such a question answered, and I don’t know that this has been done. Closest thing I can think of is the book “How Music Works” by David Byrne, where he follows the history of audio production and reproduction and how people’s listening preferences changed along with the live setups and the technology. Even deals with placebo effects briefly, showing there are real changes in the brain when liking something because of placebo that are the same ones as when liking something because it’s measurably different etc. So it seems like a good source to get a general understanding of how complicated the topic of “why people like stuff” is.
It sure changes the phase, but since the phase is changed for all the channels equally, there’s no worry.
If you have a multi-channel EQ and decide to change one channel’s frequencies, that’s where things get bad.
Whether the whole filters are audible is another discussion. But what linear phase does to the signal is worse compared to the phase shift (which shoudn’t be audible at all since all the channels are changing equally).
Question. What ladder DACs have you actually heard or is your life purely about numbers and not whether or not your music sounds good?
Plus I’m calling this out, starting out with a big post from ASR on this forum is a bit of a troll move. As someone that notices ladder DACs reproducing sound more accurately to my ears than delta sigma DACs I honestly don’t give a damn about the numbers when my sound is neutered. A song should not be messed with tonally for the sake of numbers.
1 Like
Already dealt with this above, maybe read the thread before accusing others of trolling. 
I use a DSD1793 which is similar to ladder dacs. Not the same, but still similar.
To my ears, the PCM section has problems with transition. The DSD section which acts as a low pass filter and nothing else, sounds better.
As someone who worked at Bell Labs on ASICs primarily focused on DACs and ADCs and how interconnects and pre-amps factor into the entire solution for customers on in the GHz range I understand. But the fact nobody has a done transient analysis and base everything of single sine wave measurements means someone is picking a choosing the datasets.
If you can hear something substantially different in terms of accuracy to pure analog, but the numbers aren’t backing it up (in terms of information), then there is a clear issue with the data taken in the tests or how the conclusions are being drawn out.
My Spring 2 has a terrible DSD implementation where on that unit I cannot compare DSD fairly to NOS modes fairly. DSD, PCM, and NOS kind of doesn’t really matter a terrible amount seeing how that describes how the data is related to the actual DAC portion of a DAC to go digital bitstream → DAC → analog filters/analog back end. I will say that a DAC that implements DSD properly will be technically superior and will have a higher global maxima inherently in its design than PCM for the DAC portion. However, implementation of DSD with delta sigma may still suffer transients that a ladder DAC may not experience.
Delta Sigma designs rely on 4-6 (chord uses 5 in the mojo) 1 bit integrating amplifiers that are summed together. Keyword here is integrating… Which means off the bat there is some design challenges for transients and inherent phase constraints. However, at low frequencies and removing phase questions aside will give the smooth and clean lower frequency sign wave, which is where we see all the measurements for THD. This figure has an impact on the sound but it’s only a piece of the puzzle, I can flat out look at that number and say some of my gear will not respond well and others will be fine.
Now this is were things get interesting between DSD and PCM with delta sigma approaches. For PCM the pulse train into the digital part measures in the normal 44.1k for regular CDs, based on that pulse train internally (usually up sampled to the highest sample supported to keep the digital to analog simpler) the pulse stream in converted to those 4-6 1-bit integrating amps. With PCM I think the max is like 768kHz samples I’ve seen but no clue if the D to A actually implaments that. Now with DSD, bump that up to like 22MHz for the pulse train coming in, MUCH better transient response!
As for ladder DACs, it’s simple. It’s a parallel 1-bit network of resistors without any need for capacitors. Assuming the amplifier can ensure a buffer with minimal phase to be introduced and a capacitor for the transients, it just works.
1 Like
That’s the beauty of DSD. It’s inherently linear. A lowpass filter gives the signal. No need for data-weighted averaging or anything else.
Multi-bit delta-sigma modulators bring back the problems of ladder dacs.
The bad thing is that it needs quite a bit of process in digital form if the source is PCM. But nowadays, a smartphone can do the job.
Not exactly. That is an extreme over simplification.
Uhm… Either you are phrasing something improperly or I am missing some type of reference that is missing here because the last thing I would say is a low pass filter.
With my 1 piece of equipment I can definitely say I experience VHS level compression with recessed mids. But one does not purchase a Spring 2 for DSD exclusively.
Ladder dacs have linearity problem which is one of the problems of multi-bit DS modulators, too.
This document offers DEM solution for this problem.
A lowpass filter is enough for DSD (1-bit DS modulator) to get converted to analog. While I couldn’t find what Spring II does exactly, I know Airist DAC which is another ladder dac, converts DSD to PCM.
Whatever Spring II is doing to the DSD signal is not necessary and actually damages it.
DSD1793 has a different path for DSD which is only a lowpass filter.
I’ve enjoyed reading through this thread despite the obvious defensive nature of positions established by all. It’s given me a headache and I’m a technology guy by vocation and avocation. lol
Last night I listened to the Time Out album by the Dave Brubeck Quartet. The chain was thus; Elac Discovery Streamer > Border Patrol SE > Schiit Freya + > Carver m 1.0t Mk II > customized Zu Dirty Weekend.
I didn’t think about the technology once, one way or another. Glorious.
There’s space enough for every enthusiast to find what they love and hopefully what they love comes in at a budget they can afford. As-salamu alaykum
7 Likes
To be honest, you did admit that you were not being off-topic and now this is purely proof that you interrupted our pleasant conversation and completely derailed it. You have been asked multiple times, politely, but have ignored it.
The next time I am looking for someone that uses bastardized science to prove their subjective idea, I’ll know who to look for.
1 Like
Insistence on making it personal is a great way to get yourself on my ignored users list, which until today never seemed necessary in this forum, but oh well. I guess it’s one of the signs the user base has been growing.
Hello,
I asked stupidly I have the Singxer Sda 2 advance as Dac which also has a nos function.
I notice marginal differences when switching on the Nos mode compared to S-sharp or S-low.
S-sharp I find I get more out of the top and at the end I miss it a little bit with Nos.
Does this mean that if I turn on Nos, the music will play the way it was composed?
And the filters would just make it more beautiful?
Actually, I’m not complaining because Singxer didn’t even do that badly, I’ve had worse things where you hardly noticed anything.
Or does the topic only concern the above mentioned Dacs where some of them were equipped with the Philipps chip?
1 Like
The Holo Cyan isn’t based on the Phillips chip so the subject/question is at large. If it sounds better to you though then that’s all the answer you need! There’s a lot that goes into a DAC besides the chip and whether it’s doing OS or not. In some cases the answer may be yes, in other cases, no.
¯_(ツ)_/¯
2 Likes
I’ve spent about a month now doing some extensive A/B switching with the Holo Cyan, whatever they’re doing or not doing on the NOS implementation of their R2R array. I can say without a doubt that I prefer what’s happening to the sound in NOS mode.
There’s a clear sense of the space a singer is in. A great example of what I mean is Candy Everybody Wants. Natalie Merchant’s MTV Unplugged live recording. You get a much better sense of the space she’s singing in.
I don’t have enough experience to know if that’s specific to this DAC or my chain or the fact that it’s a Monday, today. 
I’ll try to spend some time on the Border Patrol with the same songs to see what I come up with there. It will be a different chain, so not sure what value there will be to that. Just paying forward to the community here.
MusicBee (local file) > LKS DDC > I2S Holo Cyan > LP > ZMF Eikon
2 Likes
Forgive me for raining on this parade of knowledge. One point is always missing from the discussion - everyone seems to forget that signal at oscilloscope or speaker wire is not exactly what the listener would hear.
The most analog component in the universe is speaker drivers that simply cannot move from point A to point B instantaneously. It is a mechanical device and it takes time to move. This is the last step in digital to analog transformation that gives you that original sound and final smoothing of any steps side effects of digitizing you may see on the oscilloscope. The quality of the drivers do matter, but none is exception from the law of physics. This is not as important in analog-to-analog domain as there is no such steps normally present in the music.
I finally got decent R2R with NOS and started to enjoy it.