Oversampling vs. Non-Oversampling DAC

[nyhifihead]

implementation is everything... but I am a big fan of my NOS TDA1541 and PCM1704uk r2r chips.

IMO though OS vs NOS isn't as valid as the dual differential vs single chip implementation. As a source noise floor is everything, and with the PCM1704s dual differential s/n is over 130db, and in my own subjective listening tests I prefer the dark backgrounds that the lower noise floor provides

audio gd's reference dacs now even have 4x and 8x dac chip implementatiosn for even lower noise floor.. I want to listen to those

 

[asilker]

implementation is everything... but I am a big fan of my NOS TFA1541 and PCM1704uk r2r chips.

IMO though OS vs NOS isn't as valid as the dual differential vs single chip implementation. As a source noise floor is everything, and with the PCM1704s dual differential s/n is over 130db, and in my own subjective listening tests I prefer the dark backgrounds that the lower noise floor provides

audio gd's reference dacs now even have 4x and 8x dac chip implementatiosn for even lower noise floor.. I want to listen to those

was reading up on multiple chip implementations today. From what I understand, as chip technology progresses, the devices demand to be run in more ideal conditions. As a result of pickier chips, it becomes more difficult to link them together as they have been.

What I really need to do is own a few different designs so I know how they feel...

 

[nyhifihead]

was reading up on multiple chip implementations today. From what I understand, as chip technology progresses, the devices demand to be run in more ideal conditions. As a result of pickier chips, it becomes more difficult to link them together as they have been.

What I really need to do is own a few different designs so I know how they feel...

very true. regarded as the "giant killer" on whatsbest forum and with praise on audiogon/computer audiophile, is the gustard x20, its uses the famed ess9018, but in a quad dual differential configuration 4x 9018 dacs per channel in a true dual mono, and people with MBL and 10k+ high end dacs say that this dac is serious business, speaks for the implementation.

https://www.amazon.com/GUSTARD-DAC-...dp/B015R112XA?ie=UTF8&*Version*=1&*entries*=0

although in a a different price bracket.. the audio gd master series also gets very high praise.. it uses 4x pcm1704uk per channel in a dual mono configuration

http://www.audio-gd.com/Master/Master-7/M7EN.htm

I would be hesitant to get the audio gd though, as the schiit yggdrisil has insane praise in that price bracket.. interesting none the less


also those who praised the gustard x20 were mostly running the i2s protocol from a digital interface.. that is a discussion for another thread tho haha

 

[stoutblock]

I remember this old thread! Now I own an audio-gd Master 7 and love this DAC! Even running NOS it comes across full and without noticeable noise. John Coltrane's Sax work is to die for in NOS and 130db stopband. Also own a Reference 5.32 (2 x pcm1704uk per channel) and it is a giant killer IMHO.

 

[restorer-john]

There is so much regurgitated BS in relation to NOS and it makes me laugh. Most of the NOS jockeys are stupid enough to implement their braindead topologies without a decent brickwall filter (because they have read and believed a good fs/2 filter is so bad but, guess what, it isn't!) and they certainly don't FFT their creations, because actual test results mean nothing to them and yet they will wax lyrical about the wonderful subjective performance of these abominations to anyone who will listen. (ie: they are so full of ultrasonic crap they quality as a LF transmitter)

I am willing to bet there is not a soul on this site who could audibly distinguish a competently operating, 1st generation CD player (NOS 16 bit or 4xOS 14 bit) from any DAC they'd like to dream up, when fed with a 16/44 stream pulled from a commercial CD in a live comparison.

 

[SoNic67]

I remember this old thread! Now I own an audio-gd Master 7 and love this DAC! Even running NOS it comes across full and without noticeable noise. John Coltrane's Sax work is to die for in NOS and 130db stopband. Also own a Reference 5.32 (2 x pcm1704uk per channel) and it is a giant killer IMHO.

Because the NOS in that thing goes trough a true DSP that does the proper digital filtering.
Most of the NOS garbage out there has nothing for digital filtering, nothing for analog filtering. And if is a multibit R-2R you have just alias products to worry about (some even like them).
But if you manage to use a delta-sigma chip in NOS mode, the ultrasonic noise dithering will overwhelm any stage that follows them.

 

[stoutblock]

Over all I like the 8x OS on the Master 7 with stop pass filter set at 130db but there is a fun effect listening to NOS on this DAC. Like I said, Coltrane's solo Sax work is just killer in NOS.

 

[asilker]

I've said this in another thread, but I'm really looking forward to some firsthand experience with all this

 

[Gibsonian]

I need to rob a bank for new DAC funds. You guys with funds are killing me! But please keep it up.....

 

[asilker]

Hey folks. I've been spending some time with the Starting Point Systems DAC3 which is NOS R2R and features a filter less design. I have read both praise and demerit for how it sounds. I'll make my remarks after living with it for a while longer, but first impressions are favorable.

 

[dpatters]

Found this thread helpful. I'd be interested to hear about this in light of some of the recent hubbub about R2R DACs.

I've had both types of dacs in my system. I prefer the NOS for Redbook CD (I spin CDs on a dedicated CD transport) I am currently using Metrum Acoustics Musette DAC.

 

[asilker]

Here's an interesting thread from audiogon that makes a case for unfiltered NOS r2r DACs. From my understanding, going filter less has got to be implemented with specifically nos DACs, and this r2r. Anyway, here's the thread, and bombaywalla is the one who's posts you'll want to read: https://forum.audiogon.com/discussions/nos-dac-s-without-any-digital-filtering

@SoNic67 id love to hear your thoughts on this

 

[SoNic67]

I agree with what he said, maybe less about the importance of the reconstruction filter. In my experience that is very important for R-2R and essential for any Delta-Sigma DAC. I will talk only about the R-2R.
Now, that being said, he is right about the digital filters used for the OS. In my experience some of them are just bad, some are just OK... but a really great one has to be implemented by a dedicated DSP chip. A DSP can have a more refined OS algorithm (hopefully it is actually implemented).
A reconstruction filter for 44.1kHz that leaves the audio untouched is basically impossible to build because of the proximity of frequencies (20 VS 22.05kHz). Actually lots of the recordings are done at 96kHz to lessen the burden on that end (ADC).

The issue is because how a filter deals with the transients, pre and post ringing, intermodulation products and other effects (not very well understanded). Some of the filters have different characteristics (sharp roll off, slow roll off, etc), and each in the end is a compromise of some sorts between reconstruction (amplitude domain) and group delay (time domain). My guess is that some of us are more sensitive to the time errors (this is a brain function) and they would rather tolerate the imperfect reconstruction, that might be why some prefer NOS.
Actually I think that is also the reason why some (like me) like so much the sound of R-2R versus a D-S - the time domain issue that plagues the D-S conceptually.
I think that an NOS player can still be used correctly (at higher sampling rates) using in front of it a dedicated OS solution (software on computer, DSP), followed by a minimal filter - that's possible on the higher frequency, it doesn't interfere with the audio band.
But you cannot go to the extreme of no filter. The gains on time domain are completely overtaken by the damage done in the analog domain. Some amplifiers simply cannot handle the extra ultrasonics pushed in them and will generate extra IMD. The tweeters might be overwhelmed too...

Now, for older people (40+), using good headphones (that can handle the ultrasonics), the ear cannot hear those high frequencies and it acts like a natural reconstruction filter at some 15-16 kHz, far enough from the 22kHz needed. I noticed that usually older people praise the NOS solution.

PS: About the Delta-Sigma (or Sigma-Delta, whatever school of thought you like) DAC's: Read the Sabre whitepaper that I have linked here (especially about the D-S modulator), you will see that the engineers are aware of the D-S issues and are trying to mitigate them while maintaining the production costs low. "Laser trimming" of an R-2R chip is not where they want go anymore... sadly.
Just a small passage, but there are more issues covered:

ΣΔ modulators when provided with a rapidly changing input signal will exhibit non-linear noise behavior as they process the transient. This is because all noise shaping modulators are feedback systems and the usual design process (supported by commonly available design tools) operates to minimize in-band noise suppression while maintaining stability. This noise-optimized stable loop configuration will lead to an output that matches the input to the required degree within the requested bandwidth as expected. However this typical design process neglects the dynamic response of each state variable: there are choices of Q (and relative gain) that minimize noise but result in relatively large lightly damped resonances of the internal state variables. The consequence of this is that in a quiet passage of music the state variables of the modulator are all operating within a certain “state space” and the quantization noise shaping is described by the noise characteristics in this “volume” of the space. After a large music transient has passed, the output traces its dynamic response back to the quiescent operating point as we expect, but every state variable is also following its transient response back to its quiescent. During this multi-dimensional excursion back to the lower signal level the operating point traverses different volumes of the space, each of which has its own noise characteristic. Hence a very perceptive listener can hear something “anomalous” related to the transient response

Another issue is this:

it is a known problem in noise shaping modulator design that noise is not independent of DC signal level. A graph of DNR vs. DC signal level will always rise as the modulation depth approaches 100%. Most modulators cannot handle 100% modulation depth and what they call full-scale may be only 50% modulation depth, nevertheless a graph of DNR vs. modulation depth up to even this 50% level will show an increasing noise

That made a lot of manufacturers (TI is one example) go towards the combined multi-bit ( for the higher bits) and D-S (lower bits) or made others to use multiple D-S blocks in parallel, thus effectively lowering the modulation on each of them.

However for a DSD (SACD) signal there is nothing you can do to "fix it". You can go higher in frequency (DSD128, 256, 512), pushing the garbage further under the rug.

 

[asilker]

I agree with what he said, maybe less about the importance of the reconstruction filter. In my experience that is very important for R-2R and essential for any Delta-Sigma DAC. I will talk only about the R-2R.
Now, that being said, he is right about the digital filters used for the OS. In my experience some of them are just bad, some are just OK... but a really great one has to be implemented by a dedicated DSP chip. A DSP can have a more refined OS algorithm (hopefully it is actually implemented).
A reconstruction filter for 44.1kHz that leaves the audio untouched is basically impossible to build because of the proximity of frequencies (20 VS 22.05kHz). Actually lots of the recordings are done at 96kHz to lessen the burden on that end (ADC).

The issue is because how a filter deals with the transients, pre and post ringing, intermodulation products and other effects (not very well understanded). Some of the filters have different characteristics (sharp roll off, slow roll off, etc), and each in the end is a compromise of some sorts between reconstruction (amplitude domain) and group delay (time domain). My guess is that some of us are more sensitive to the time errors (this is a brain function) and they would rather tolerate the imperfect reconstruction, that might be why some prefer NOS.
Actually I think that is also the reason why some (like me) like so much the sound of R-2R versus a D-S - the time domain issue that plagues the D-S conceptually.
I think that an NOS player can still be used correctly (at higher sampling rates) using in front of it a dedicated OS solution (software on computer, DSP), followed by a minimal filter - that's possible on the higher frequency, it doesn't interfere with the audio band.
But you cannot go to the extreme of no filter. The gains on time domain are completely overtaken by the damage done in the analog domain. Some amplifiers simply cannot handle the extra ultrasonics pushed in them and will generate extra IMD. The tweeters might be overwhelmed too...

Now, for older people (40+), using good headphones (that can handle the ultrasonics), the ear cannot hear those high frequencies and it acts like a natural reconstruction filter at some 15-16 kHz, far enough from the 22kHz needed. I noticed that usually older people praise the NOS solution.

PS: About the Delta-Sigma (or Sigma-Delta, whatever school of thought you like) DAC's: Read the Sabre whitepaper that I have linked here (especially about the D-S modulator), you will see that the engineers are aware of the D-S issues and are trying to mitigate them while maintaining the production costs low. "Laser trimming" of an R-2R chip is not where they want go anymore... sadly.
Just a small passage, but there are more issues covered:


Another issue is this:

That made a lot of manufacturers (TI is one example) go towards the combined multi-bit ( for the higher bits) and D-S (lower bits) or made others to use multiple D-S blocks in parallel, thus effectively lowering the modulation on each of them.

However for a DSD (SACD) signal there is nothing you can do to "fix it". You can go higher in frequency (DSD128, 256, 512), pushing the garbage further under the rug.

This is a very content rich post, I've read it through a few times and will continue to in order to absorb it better.

My amplifier is fully tube. Doesn't the cap coupling allow the first tube stage to act as a limited band-pass filter and thus eliminate ultrasonic images?

 

[SoNic67]

Yeah, but is just a very slow slope/roll-off - RC combinations are first order filters at -6dB/octave. To get to -60dB attenuation you need 10 octaves - that is a huge delta (2^10 = 1024 times the frequency). The Digital Filter is very important there (DSP).
Or your tube amp ,that can handle MHz at the input, has also transformers that kill a lot of ultrasonics...