...DSD512 should be no better than your current PCM audio on an ESS DAC...
What should be and what is are not always the same thing. Experimentally, some modern dacs sound subjectively better in DSD mode than they do in PCM mode.
I just came across a posting with this unconventional Denon Alpha Dac with a continuously switching sharp/slow FIR filter, depending on the incoming signal.
No idea what problem should be solved with this solution, and obviously they didn't conquer the market with it.
But always nice to have a glimpse on what's being tried, possibly leading to new ideas.
The Black Hole......
Hans
P.S. Alpha stands for Adaptive Line Pattern Harmonized Algorithm.
Edmund Meitner, founder of museatex and Meitner Audio, patented the attached similar concept in 1995. The general notion of both, seems to apply analog signal theory to digital signal implementation. Even though, they are both well regarded for their digital systems knowledge. Such notion being the view that SINC-function based FIR filter impulse-response ringing is undesired for reconstructing a sampled signal. That was probably subjectively true, except not for the reasons which they apparently suspected.
As we just found out in our investigation, high-performance sharply bandlimited, long-ringing FIR filter playback, essentially sounds like zero-ringing NOS playback. Suggesting that Denon's Alpha DAC and Meitner's patent were, unknowingly, intended to compensate for audible artifacts introduced by common OS interpolation-filters.
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That doesn't mean anything to readers unless measurements can be presented for comparison because it may have been all in your imagination.Experimentally, some modern dacs sound subjectively better in DSD mode than they do in PCM mode.
We just found that some of our investigators were unaware how NOS playback really sounds and unknowingly draw conclusion that NOS HF roll-off must be compensated, otherwise is heard. A final conclusion is flawed, but not surprising in this situation.
I do support @Markw4 statement. Some Delta-Sigma DAC's sound better in DSD mode indeed.That doesn't mean anything to readers unless measurements can be presented for comparison because it may have been all in your imagination.
DSD is largely unnecessary in the modern world. Perhaps it had benefits when 25MHz was considered significant, but these days it's easily traded with equivalent PCM, e.g. DSD512 should be no better than your current PCM audio on an ESS DAC. Modern sigma-delta DACs work similar to the DSD principle with generally a few more bits, maybe 4 or 5. The age of single-bit sigma-delta converters, which would be true to DSD, and their spurs/harmonics, are over. Their inherent linearity is not worth the downsides when a 2nd-3rd order sigma delta ADC is easy to design, and a 4th-5th order sigma delta DAC is also easy to design thanks to modern digital simulation.
Either you are mixing up feedback loop orders and wordlengths, or I don't understand you, or both.
I think the reason why multibit sigma-delta modulators got popular is mainly the invention of mismatch-shaping algorithms that help to get a decent performance out of a multibit DAC without insane matching requirements.
Single bit can also work well when you use the old trick of embedding a pulse width modulator in the sigma-delta loop to make it a quasi-multibit system. The quantizer can then be multibit and can hence be dithered according to dither theory.
With sigma delta, and oversampled SAR/R2R/etc. converters, the trouble always comes back to filtering. How well is the input data filtered during the conversion from PCM to "DSD-like" data or, for both, interpolated oversampled data? How many taps in the filter? Is it a simple digital sinc-in-frequency (SincX) filter? Well, that's not nearly good enough because an ideal filter is sinc-in-time, which is much more computationally expensive to approach, and perhaps impossible to reach (causality issues).
This is the same problem that early CD player designers had. Referencing the SAA7220 - it was a great achievement of the era! But 120 taps of an FIR filter is absolutely pathetic these days. I could easily drop a 10000 or 100000 tap FIR filter on a digital audio file in real-time (obviously delayed, but still streaming) now. 120 taps is nowhere -near- being sinc-in-time, and likely sounds bad. Note that chip's datasheet has -zero- information about its added THD/SNR/etc.
The only things that can generate distortion are clipping on intersample overshoots and roundings of intermediate results and rounding of the end result. My former colleague Frans Sessink suspects that the SAA7220 rounds intermediate results. The digital filters in the first Philips CD players certainly rounded the end result with first-order noise shaping and no dither.
As long as there is no clipping or folding and you don't round anything, an FIR filter that interpolates by an integer factor is linear and time-invariant and can't distort, no matter how short or long it is. It can have a crap frequency response and pre- and post-echoes, though.
What would you do or have you done to confirm that it wasn't all in your imagination? Any measurements or objective comparison data to share?I do support @Markw4 statement. Some Delta-Sigma DAC's sound better in DSD mode indeed.
Edmund Meitner, founder of museatex and Meitner Audio, patented the attached similar concept in 1995. The general notion of both, seems to apply analog signal theory to digital signal implementation. Even though, they are both well regarded for their digital systems knowledge. Such notion being the view that SINC-function based FIR filter impulse-response ringing is undesired for reconstructing a sampled signal. That was probably subjectively true, except not for the reasons which they apparently suspected.
As we just found out in our investigation, high-performance sharply bandlimited, long-ringing FIR filter playback, essentially sounds like zero-ringing NOS playback. Suggesting that Denon's Alpha DAC and Meitner's patent were, unknowingly, intended to compensate for audible artifacts introduced by common OS interpolation-filters.
Thx for this link.
Reading the patent a solution was proposed to solve the following problem:
Pre ringing distortion is highly audible in music passages and is the major audio difference between analog processed playback and digital playback.
As you already mentioned, this is not the case.
On top of that one should be aware that pre ringing may already start much earlier where a high Fs digital recording is downsampled to 44.1 using the same sort of brick wall filters as used for OS.
Whatever smart type of OS filter will be used at a later stage in the DAC, this filter will never be able to remove eventual pre ringing in the available music content.
Feeding a Dac with a dirac pulse to visualise things may therefore be misleading.
Hans
If something sounds better than the other, it would first have to sound different. Confirming that can be done by comparing measurements and or level matched double blind listening test.How do you propose showing an objective comparison of sounding better?
@ Even,
I read hundreds of posts (at least that is what it felt, did not really count) where you keep nagging on measuring etc and trolling about sound quality perception.
so I took a look at the threads you started. Do I need to say more? Specially the thread on the Ebay amplifier and the first reaction to that made me start the day with a smile.
You are really a fun boy, keep presenting everyone as the man who gets it all in other threads. Really funny - I should have done this earlier, I take your post not so seriously anymore 😀
I read hundreds of posts (at least that is what it felt, did not really count) where you keep nagging on measuring etc and trolling about sound quality perception.
so I took a look at the threads you started. Do I need to say more? Specially the thread on the Ebay amplifier and the first reaction to that made me start the day with a smile.
You are really a fun boy, keep presenting everyone as the man who gets it all in other threads. Really funny - I should have done this earlier, I take your post not so seriously anymore 😀
If something sounds better than the other, it would first have to sound different. Confirming that can be done by comparing measurements and or level matched double blind listening test.
OK, if you want to play that silly game, let's presume a difference has been proved, now how do you answer the question? Do you see yet the meaninglessness of your question? Your questions in this vein are unanswerable as I'm guessing you already know, so, pray tell, why do you keep asking them?
Just because it's unanswerable to you doesn't mean it's the same for sajunky (whom I asked what he would do to confirm one's observation is all in his head or actual audible difference). Perhaps you should've consulted with sajunky first.Your questions in this vein are unanswerable
Since you are new here, some background info is what you need. Try this, Audio Myths Workshop - YouTube
The question was about better, we'd presumed a difference had been established, please try to keep up, how would one show with data that something sounded better? PS, I don't expect an answer so please don't try too hard.
As I suspected, you are new to this forum and the world of sound reproduction as well. The quality of sound reproduction equipment is judged on its level of fidelity, higher the better. If you aren't familiar, look up the term hi-fi (high fidelity). Something sounding better in reproduction would have higher fidelity to its reference source. I suspect that you thought it was something a listener likes better. That would be a valid criteria if you are judging a musical performance but this thread is about audio gear for reproducing such performance. Try to understand the difference between the two.
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