Why does the soundstage sound different (often better IMHO) in high rate DSD like DSD256 Vs native Redbook to a DAC with a Chip that upsamples to ultimately do SDM conversion.

[Miska]

18 minutes ago, Shadorne said:

Good point. This difference in digital processing could be the problem. However, we are talking about devices that measure extremely well. All modern devices using processing on a chip or PC have distortion at levels that is believed to be inaudible. To affect the soundstage the distortion obviously must be audible. Since devices measure so well, it must be an extremely low level distortion in time but somehow audible - this is hard to believe but it must be true or why else would so many agree that there are audible differences between the two approaches in processing.

 

Or wrong things are being measured... Or the things aren't inaudible in the end...

 

I have my own opinion on these matters...

 

18 minutes ago, Shadorne said:

I postulate it is the pre-echo from an equiripple filter which is typically found in chip based processing. Pre-echo is an exact copy of the signal that occurs around 0.5 millisecond early but at extremely low levels whenever a filter has a constant ripple in the passband.

 

Pre-echo is not related to pass-band rippleor equiripple, it is just side-effect of high order linear phase filter. Longer the filter, longer it rings. Chord being extreme example.

 

18 minutes ago, Shadorne said:

The effect, if audible, would cause the ear brain to determine that the sound came from the direction of the speaker. The ear/brain would be confused by this pre-echo as it conflicts with timing information of the stereo image (which also has distinct time arrivals of the audio that create the illusion of position of sound within the soundstage).

 

Does anyone know if there is passband equiripple in HQplayer or Roon computer based digitally processed signal and how does this differ from typical equiripple from the simpler processing on a typical DAC chip?

 

Well, this has nothing to do with pre-echo. But for example typical DAC chip filters have pass-band ripple of around 0.01 or 0.001 dB. HQPlayer filters have typically pass-band ripple below 0.000000001 dB.

 

 

But there are many factors, and only 50% of the performance comes from the digital filters. The other 50% comes from the modulator design.

 

[Miska]

9 hours ago, STC said:

5) The question is - How higher or lower resolution would alter that. 

 

It is both time difference and level difference. Plus you have a very complex waveforms with lot of harmonics.

 

As the analog output signal becomes more precise, it becomes easier to distinguish the different harmonics and their timing differences.

 

[Miska]

8 minutes ago, Shadorne said:

Pre-echo or post-echo from bandpass ripple is not the same as ringing. I am not referring to ringing. Ringing is from the brick wall filter or boxcar effect or Gibbs phenomenon. I refer actually to the distinct echo from the ripples in the frequency domain of the passband.

 

Those are effect of Gibbs phenomenon (step- or impulse response). In HQPlayer you have number of filters as both linear- and minimum-phase variants where you can compare the effect. These are otherwise exactly the same, but minimum-phase filter has no pre-echo, just post-echo. Same pass-band ripple.

 

You are talking about pass-band ripple which is a separate parameter, just like stop-band attenuation and transition band width. Plus of course gazillion of other parameters. Pass-band ripple is just frequency response variation and doesn't cause any "echo".

 

[Miska]

2 minutes ago, STC said:

For ILD and ITD, the brain expects a set of response to match what the other ear heard for localization. The experiment is easily conducted to be prove the point.  Pick or make a DSD recording with an object about 20 degrees off center. Perhaps a speech. Split the left and right channel. Down sample just one channel to 44.1 khz. Introduce a little noise that is said to be produced by the lower sampling rate. Now mix the lower sampling rate to the other untouched DSD channel. So you have one channel of 44.1 and the other a pristine DSD.  The 44.1 need to converted to DSD. Now let’s listen and see if the image changes position.

 

That doesn't have same effect as running DAC at lower sampling rate...

 

If you like, you can simulate the DAC errors in digital domain for one channel. But you need to do that at over 10 MHz sampling rates.

 

[Miska]

Just now, Shadorne said:

According to Julian Dunn, equiripple in passband does cause distinct pre and post echoes. 

 

Pre-echoes caused by linear phase filter pre-ringing is several orders of magnitude bigger.

 

Just now, Shadorne said:

My observation of differences in soundstage (Redbook Vs pc-based DSD256 with a true 1 bit SDM DAC) has led me to conjecture that the equiripple in the passband might be the audible problem with almost all DAC chips.

 

Well, HQPlayer has that covered at least, sot that it is non-issue.

 

But in my opinion, there are other bigger factors... Not limited to just digital filters.

 

[Miska]

3 minutes ago, STC said:

I would appreciate that but why 10 MHz? But if you can create something like that and could hear the difference than there could be something more than ILD and ITD because I suppose you can measure exactly the ITD and ILD .

 

Because we are discussing effects in the analog waveform reconstruction.

 

[Miska]

16 minutes ago, STC said:

Okay. Anything will do as long you can demonstrate there is a shift. 

 

What shift?

 

[Miska]

24 minutes ago, STC said:

 

 

 

 

If you are curious, you can do such experiments as much as you want. But you need to be careful to simulate the DAC effects exactly.

 

[Miska]

5 hours ago, Shadorne said:

This is a bit off topic (more maths) but the way I understand it, the pre-echo and post echo is a consequence of the finite tap filter length.

 

Which echo are you now talking about. Ringing is function of filter length. Pass-band ripple is unrelated to filter length (number of taps).

 

5 hours ago, Shadorne said:

You have a discontinuity at the beginning and end of the multiplication for each step of the integration process of the convolution. As the tap filter slides across the input signal (the convolution process) the input signal “reflects off” the end (post echo) and beginning (pre echo) of the tap filter. 

 

So more taps (steps) you have, longer it rings.

 

5 hours ago, Shadorne said:

A longer tap filter is going to have a smaller reflection than a short tap filter  (basically the amplitude is linearly related to tap filter length).

 

No, the opposite.

 

5 hours ago, Shadorne said:

This may be why Rob Watts likes the sound of super ultra long filters and refers to “smearing” in his presentations. (Because there is a linear relationship you have to go to enormously long filters to get a meaningful reduction in echo amplitude)

 

His filters ring for about 700 ms. So you have 350 ms pre-echo and 350 ms post-echo.

 

5 hours ago, Shadorne said:

I would expect apodizing might also be used to soften the echos as well as other tricks but inherently apodizing changes the overall result so it has fidelity issues itself (which some may argue are inaudible if done properly)

 

Apodizing exists to deal with problems originating from ADC digital filters. And possible rate conversion methods used as part of the production chain.

 

 

[Miska]

6 hours ago, Shadorne said:

Sorry to be pedantic but this isn’t an echo at all it is ringing.

 

No, you are wrong. It is echo, the filter taps before the main lobe affect output of the filter before the actual event in time happens (pre echo). It is kind of time machine achieved through equivalent amount of delay. And the taps after the main lobe affect output of the filter after the actual event (post echo). More taps you have, longer it echoes.

 

6 hours ago, Shadorne said:

Pre-ringing isn’t audible unless your hearing is good at that highfrequency which is usually between 20KHz and 22 KHz.

 

It is at, or very close to audible band that is why it is a problem. Even more so when you use for example linear phase EQ, this is why most room EQ is minimum or mixed phase.

 

6 hours ago, Shadorne said:

BUT - and this is where I am going - can someone test to see if this could be the mechanism by which we hear a major difference when upsampling Redbook to DSD64 or higher on a powerful computer with a high degree of precision and accepting the related high latency (which all DAC designers try to avoid).

 

If you compare number of HQPlayer algorithms, there are still differences although the pass-band ripple is not a concern.

 

Yes, it is one of the many aspects that affect the result, but there are other, bigger ones.

 

6 hours ago, Shadorne said:

It would be simple enough, if HQplayer offered an upsampling option that mirrors what goes on in various popular chip based DACs like ESS, Wolfson, Cirrus, and TI/Burr Brown - then users could compare directly - even A and B. Audibility could be confirmed to be processing based.

 

As I said above, you can test the theory in other way too. Compare various HQPlayer algorithms that still have the passband ripple well below anything that analog domain can represent.

 

But it is not all attributable to one single digital filter property, it is much more complex.

 

You can also compare apodizing filters (most HQPlayer filters) vs non-apodizing filters. Since you have such errors, among with others embedded in the source content from the ADC and production chain filters. Apodizing filters can correct such errors while non-apodizing will reproduce the source content errors too.

 

[Miska]

My point is that there is no one single parameter that would be a measure to cover the difference.

 

[Miska]

2 hours ago, Shadorne said:

And yes an apodizing filter would be intended to reduce that echo, although at the expense of accuracy of the transient response.

https://www.nanophon.com/audio/antialia.pdf

 

refer to Section 2 in the above paper.

 

No, apodizing filters improve transient response by fixing such errors among others in the source content.

 

I cannot see anything related to apodizing filters in that paper.

 

[Miska]

7 hours ago, fas42 said:

Yes, the 'wrong things' are what are always measured - which is why it's typically such a torturous path for people to evolve a system into producing highly convincing playback.

 

Not always, I believe I measure number of relevant things.

 

7 hours ago, fas42 said:

Again, noise and interference are the enemy; 

 

That is something that can be measured pretty easily.

 

[Miska]

2 hours ago, Shadorne said:

Section 2 is about bandpass ripple and pre-echo and post echo.

 

That has nothing to do with apodizing filters. HQPlayer for example contains multiple filters in apodizing and non-apodizing form where passband ripple values are exactly the same.

 

2 hours ago, Shadorne said:

I agree apodizing can soften “errors” produced by the discontinuity at the ends of the tap filter. It does however change the transient response peak and width - so there is some smoothing or alteration of the shape/width of transients. So it fixes some things but not without a little trade off. 

 

No, you are wrong, apodizing filters are not related to any discontinuity errors. And of course any proper filter doesn't have any such discontinuity errors, apodizing or not.

 

There is alteration of the transient response due to the errors embedded in the source data. Once these errors are fixed, the transient response becomes corrected. So you need to fix the source data first in order to correctly reproduce transients. Those errors destroy the transient response if they are not corrected.

 

HQPlayer contains detection for these errors, so you can see when apodizing filter is needed. When you use it for content that doesn't need apodizing filter, there is no alteration either since there's nothing to fix. The only alteration is error correction.

 

So there is no harm using apodizing filter for source content that doesn't need one. But there is harm when not using apodizing filter for content that would need one. Without apodizing filter, such erroneus content is reproduced incorrectly in respect to the original signal that once entered ADC.

[Miska]

5 hours ago, Shadorne said:

Section 2 of the link I provided is EXACTLY WHAT I am referring to when I mentioned pre-echo and post echo. I realize the word echo has all kinds of other/different meanings to you so I just wanted to clarify once again what I meant is what Julian described in Section 2. 

 

This will be my last attempt at clarification, as I think the discussion seems to have drifted and I apologize for being unable to get my points across succinctly enough for you to grasp.

 

My point is that the difference doesn't solely attribute to any such single thing. Just one of the many factors contributing to the end result.

 

[Miska]

4 minutes ago, fas42 said:

I'm sure that there are hints in what conventional measurements give us which indicate issues - however, I have not read anything which neatly ties the normal numbers which can be obtained, to what is heard.

 

 

Well, that's news to me . I don't think I have read anywhere where someone has done research on the resistance of a complete audio chain to noise/interference, as regards the impact on subjective SQ ...

 

Then you haven't been reading my postings much...

 

Things like NAA also exist for a reason, which is more than one.

 

[Miska]

5 minutes ago, fas42 said:

IOW, exactly where most people are ... there is a lack of knowledge on what precisely to measure, so that one can say that the presentation passes a certain standard of acceptability.

 

No, there's no lack of knowledge.

 

[Miska]

Type of the digital filter, digital filter ratio and modulator design...

 

[Miska]

7 hours ago, fas42 said:

IOW, if you were only able to extract a set of measurements, and never actually hear, listen to the playback, you could confidently predict whether you could live with the SQ, or not - yes?

 

You were talking about noise and interference.

 

[Miska]

7 hours ago, fas42 said:

Can you measure this, or not?

 

Yes I can, it is not a problem. Noise/interference is pretty straightforward to measure.

 

[Miska]

On 10/14/2023 at 9:39 PM, Archimago said:

Personally, I feel that once I control the volume, upsampling to DSD256 (with flat frequency response) sounds more like just straight high-quality PCM playback. DSD64 with its elevated noise seems to add a subjective sense of "space" for me more than DSD128 or DSD256.

 

Do you use a DAC that converts both PCM and DSD to analog bit-perfect?

 

Have you compared with various different DAC architectures and compared how they behave at different rates?

 

I personally wouldn't bother to feed DAC at DSD64 rate.

 

With ESS PRO chip models you have massive DAC-to-DAC variations depending on the I/V stage design and how the 4+4 channel summing is done. Plus the analog filter design, plus how the DAC chip is clocked/configured.

 

[Miska]

57 minutes ago, Archimago said:

I've had listening tests with many AKM DACs using DSD-Direct over the years as well beyond the ESS chips.

 

These are naturally not native PCM, so you more like compare modulators (if you used same digital filter for both).

 

57 minutes ago, Archimago said:

Yeah, I prefer not to listen to DSD64 either... Just an observation that subjectively it made more difference than DSD256

 

For example the new AK4191 + AK4499EX chip combo always runs either at 128fs or 256fs conversion rate. With careful selection of settings this combo can also do great DSD Direct conversion at DSD128 or DSD256. Then also the modulator rates can match.