DSD vs PCM resolution

[YashN]

It's a fun domain of thought the differences between these two.

 

My POV is that DSD is really like the analogue sound pressure when sound travels in air. Case in point, if you output it on a pin, then you need only add a low-pass filter after it to hear sound.

 

Which means, you can suppose the D/A occurs on the pin output. It's a format defined in the Limbo between digital and analogue or it's analogue within a digital format since the density of 1s make for the high signal.

 

Now, another thing you could take into account is the totally ming-boggling amount of information in attack transients and try to see which format captures those better. I think the filtering at ADC with PCM is audible and probably detrimental. No such filtering on capture with DSD.

 

Additionally, when most PCM is output, there's a few more steps before output as SDM ultimately compared to native DSD output, so maybe with PCM we get more 'digititis' or more of the digital circuits' signature.

 

We can take into account the more gentle filtering with DSD on output as well.

[YashN]

An early DAC design for RB CD by Sir Clive Sinclair only needed output cable capacitance for filtering.

 

Interesting even though at the time of the Sinclair computers, I was using Oric and Commodore.

[YashN]

And to make it really clear once more: nothing else than acoustic music can ever be a proper reference in any true HI FI comparison between products.

 

I agree, technically acoustic is the only way to even have a proper reference and try to reproduce it.

 

We can't easily do the same thing with electronic or amplified music, since it's too equipment-dependent and one would need the exact same equipment to do a real Hi-Fi reproduction - not easy.

[YashN]

- my prototype was picking up lots of noise. I know what you mean about birdies. But now that I have settled down on the circuit layout, I have re-done all the wiring and got rid of a lot of 'antennas" that were the result of an untidy build. At megahertz, layout is critical. The noise is gone with a proper layout.

- I use a flip flop with complementary outputs for a balanced signal. Perhaps this helps with noise. My motivation was simply that I have a balanced system, and I wanted to keep this new source balanced as well. But I think that noise rejection may be another benefit.

- Forget Foobar. I used foobar for years, but I found HQPlayer a few months ago and there is no going back.

- I am now using a JLSounds USB card. With Linux (Ubuntu Studio 14) and HQ Player I can play DSD256. The only LPF is the capacitance in the Lundahl transformer.

 

Yep to all the above, Hazard, and great thread over at diyaudio which became much better than my older one.

 

The so-called 'birdies' most probably had to do with the implementation since the down-conversion to 88 PCM was OK, but some people conclude the format isn't good...

[YashN]

Current audio systems (that I know) can't made full reproducing of concert hall. Due there need re-create full wave picture with correction to listening room. It is not amplitude or phase response matter.

 

I think we can approach it with DSP in the PCM domain. However, nowadays I am not too fond of any DSP treatment for my music, I'd rather use high-rate DSD with gentle filtering.

 

We're thinking of going to see Kent Nagano to calibrate our ears for a fuller orchestra than what we heard in Toronto with Peter Gabriel's New Blood Tour Live.

 

I am especially interesting in noticing where the system fails in reproduction in this case, and finding organic and acoustic ways of making things better.

[YashN]

I downloaded the Trial of HQ Player. Sorry, I have never encountered such an infuriatingly unusable piece of software before. Usability scores on one out of ten are minus infinity. I pass, thank you.

 

A great way to justify not trying what is possibly the highest SQ player currently available...

 

There is an initial learning curve, but once you're past it, it's really easy to use.

 

Use PCM if you can't use DSD.

[YashN]

It may be we hearing the problems predicted by Lipshitz/Vanderkoy in their paper criticising DSD.

 

Wasn't that paper thoroughly debunked by another paper?

 

JS observed that the Burr Brown style analogue FIR Filter (as also found in several DIY DSD DAC's) do a good job of scrambling these tones out of existence.

 

I thin purely digital modulator (e.g. the one in the DSD converter from Foobar2k) can perform near the theoretical limits and avoid creating these tones. Real modulators in analogue hardware seem to have problems though.

 

There are excellent modulators and filters in HQ Player. Ideally, for further isolation, HQ Player should be used with an NAA (network-attached audio device) running a small HQ Player networkaudiodaemon but you can still get great results with a direct connection.

 

Anyway, birdies were only observed playing SACD Rips, not with (CD Standard) PCM converted to DSD.

 

Knowledge of the exact provenance could be helpful here: if the SACD was made from PCM tracks, this is going to give you vastly different results than if the SACD was made from pure DSD recordings as any conversion is lossy.

 

Besides, SACD is DSD64 only + DRM. My interest in DSD64 waned as soon as I heard DSD128, though I still look for DSD64 if I can get some (and not with DRM or SACD iso form, if I get the latter I'll extract the individual tracks).

 

For these I use the Foobar2k DSD-PCM converter to 88.2kHz/24Bit. It sounds no worse than any notional DSD DAC I had around to play with.

 

It's important here to note which DAC and how they work internally (any internal conversion and how).

 

I found no sound difference between J-River and HQ Player, non.

 

There are people with vastly better equipment than me who would agree to strongly disagree with you about what they perceive, but so long as you're happy, go with what you like.

 

If it actually sounded better I might be willing to put up with the interface (read lack thereof), lack of usability and learning curve and all if it did anything for sound.

 

Ensure you configured the software correctly (preferences and DSD settings, post a screenshot of these to get help or advice), and then play with the different modulator and filter settings, preferably with a well-implemented native DSD DAC (like the $190 iFi iDSD Nano giant-killer).

[YashN]

Debunked? L & V presented a further paper in replay on the same day that thoroughly "debunked" the papers of Philips and Angus criticising theirs, not just making claims (like the papers criticising theirs), but showing the full math.

 

Not just debunked, thoroughly debunked. Not sure if it was 'Philips and Angus' though and wish I could recall the other paper or the authors.

 

 

I only play PCM without digital filter, I fail to see relevance.

 

The relevance is obvious if you really want a good way to listen to DSD...

 

 

 

I think the SACD's I like are made from analogue master tape.

 

Maybe, but then you would prefer DSD128 or Quad DSD.

 

I had the iffy stuff here, both model (so-so sound, not much resolved, plus not balanced out which is a pain), several China made "Sabre" DAC's, a Weiss Sabre DAC (they all sounded rather similar - not bad not good) and a DIY Unit with discrete DSD DAC from Japan.

 

So you make your opinion of a format with iffy equipment? How about trying better ones?

 

My equipment is very modest. I follow the ideas of Arthur Salvatore in the system setup, due to trying his advise I removed my Tube Preamplif and got my speakers.

 

There's nothing wrong with modest equipment and I think one can make modest equipment sound really good with effort. I do like what I read a few years ago on Salvatore's site, especially when it comes to getting new equipment or improving on manufacturers' commercial items.

 

I use Thinkpad plus M2TECh hiface, Win8, Fidelizer and J-River (tried J-play, no difference). This replace CEC TL1 Beltdrive CD Transport.

 

I'd recommend HQ Player over JRiver any day.

 

1) Play CD-File converted to DSD64/128/256 via Le DAC FlipFlop, no birdies, okay sound, but not as good as Pass D1

2) Play SACD Rip file directly via Le DAC FlipFlop, birdies, strange sounds

 

Theory after observation:

 

Before embarking on any theory, what's the precise SACD title, how was it ripped, in what format, what's the Le DAC FlipFlop doing internally, can you record and share a segment of the 'birdies', etc...?

[YashN]

Very cool. I asked Bruce if he could point me to the files.

 

I think there were Ella - 'Black Coffee' and Crosby, Stills, Nash (perhaps Young) and a third one.

[YashN]

And in any case, based on somewhat recent study, the temporal resolution of human ear exceeds Fourier uncertainty principle.

 

Thoroughly interesting, Miska.

 

I liked the mention of the top-acuity guy as being an electronic musician - makes you wonder (as I sometimes do) about the very fast attack transients that are available in some analogue synths and analogue modular synths on top of their already quite large frequency range for some.

[YashN]

Filter ringing causes time domain spread of a transient.

 

And messing up transients messes up everything greatly: localisation, timbre accuracy, rhythm/timing, soundstage, presence, realism, space, etc...

 

I sometimes do this test of timbre perception on my Kurzweil K2500XS + KD/FX synth with willing visitors: I chop off just the attack of a sound of a very well-known acoustic instrument (yes, that Kurz is a DSP-based synth so it can do samples + real-time processing rather than Physical Modeling), but people don't recognise the sound anymore at all (I do).

[YashN]

I am supposing (but ready to be told I'm not correct, since I cannot presume to speak for Miska) that we must take time domain effects into account as well, and that Fourier analysis specifically may not adequately model the balance between what the time domain contributes to realism and what the frequency domain contributes.

 

The interesting thing to me was a model which had as assumption a linear system. The cochlea being a highly non-linear one, the assumption didn't hold for the model.

 

Non-linear and chaotic systems are highly interesting, and furthermore, as humans, we have a tendency to consciously try to predict in linear fashion, which is why many people have a tremendous amount of difficulty predicting technological progress as Ray Kurzweil does (an exponential model).

[YashN]

Kurzweil prediction for 2009: "Sound producing speakers are being replaced with very small chip-based devices that can place high resolution sound anywhere in three-dimensional space."

 

How are you enjoying yours?

 

Maybe I could build one.

[YashN]

Speakers array no bad way create sound wave in point of room. But impossibly manage reflections from room's surfaces.

 

Here need array of isotropic wave sources (like natural) that able manage radiation in all directions.

 

Speaking of speakers, I have grown rather disillusioned with our usual speakers which need to be positioned to perfection only to have a small sweet-spot, usually shared by a single person.

 

I think the only way I'll get out of this is to make my own Omni-directional speakers (and no built-in cross-overs) in an active XO setup.