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.

[bogi]

6 hours ago, Shadorne said:

PC-based Redbook to high rate DSD  is so widely reported to improve soundstage that there must be a distortion mechanism in all DAC chips

 

I think the answer lies in less distortion and thus finer resolution of audio data which enters the D/A conversion stage when you use HQPlayer upsampling.

Just compare 2 paths, assuming possibility of direct DSD path possibility in the second one:

 

1) Audio content at 44.1k -> Player output 44.1k -> DAC chip 1st oversampling stage (filtered, for example 8x) -> DAC chip 2nd oversampling stage (unfiltered, usually ZOH, for example 16x) -> DAC chip delta sigma sigma modulator -> DAC chip D/A conversion stage -> analog signal

 

2) Audio content at 44.1k -> HQPlayer output DSD256 -> DAC chip D/A conversion stage -> analog signal

 

In 2) HQPlayer does 1 stage (for example non 2s poly-sinc filters or sinc-L) or 2 stage (-2s filters, gauss family...) filtered oversampling up to target fs, folowed by delta sigma modulation.

 

So the difference lies in what is happening before the D/A conversion stage. In the HQPlayer case it is:
- Higher precision of all calculations. HQPlayer does all in 64bit or some critical parts even in 80bit floating point precision. Higher precision allows more operations without rounding error affecting the required output resolution. DAC chips perform all calculations in limited resolution of their fixed point format.

- Higher computer power allows to run much higher number of computing operations during the limited time between 2 output samples (1/target_fs). So pure computer power allows to get higher precision result because algorithms used are not so much restricted by number and complexity of operations they can use.

- Better algorithms. Higher CPU power allows to run more advanced algorithms. Implementation quality is very important, not only computer power. So far I was experiencing only Saracon converter output giving similar quality conversion result as HQPlayer is able to provide in real time. But most probably Saracon does not provide so many filter and modulator options and AFAIK it is intended for offline conversions only. All other PCM to DSD conversion tools I tried (foobar2000, JRiver, mansr's SoX adaption, Tascam HiRes DSD editor) provide clearly lower level of output quality. Maxim Anisiutkin's foobar2000 solution provides quite nice result considering low computer resources it uses, so it is suitable as free solution for low power environments like Windows mini PCs, tablets etc.

 

All these points lead to finer resulution (lower distortion) of HQPlayer processing result in comparison with DAC chip path result. In DAC chip, the first oversampling stage runs in repeated 2x oversampling steps, where only the final 1st stage result at intermediate fs (between two stages) is filtered, but of course by simpler and lower precision algorithm. Every processing step introduces some level of signal distortion. Then the fine output resolution is further restricted by 2nd oversampling stage by 2 aspects. The first one is extremely simplified sample rate calculation (ZOH or linear interpolation) and the second one is missing filtering, resulting to audio band images present on 2nd stage output at multiples of mentioned intermediate fs. Such repeated audio band images are not present in HQPlayer DSD output. Summing up, digital signal, which enters DAC delta sigma modulator, was processed by lower quality oversampling algorithms and contains unwanted ultrasonic content, correlated with audio band, which may become source of intermodulation distortion and influence downstream equipment. That content then enters DACs delta sigma modulator. That circuit quality is of course again restricted by DAC chip hardware resources. Often it is 3rd order modulator, not in pair with HQPlayer modulators.

My understanding based on listening experience is that higher precision processing result leads to better sense of air and space, clearer instrument placement and separation, better layered soundstage (instead of flat), finer and more detailed transient presentation instead of typically hardened PCM path transient presentation, fuller and more realistic instrument timbres, better dynamics because of lower noise floor, better audible low level audibility and detail of instruments like percusion etc.

[bogi]

6 hours ago, STC said:

So how this values can be changed using higher sampling rate?

You need to assume existence of two oversampling stages in DAC chips. Oversampling as such ends at delta sigma modulator rate, which is typically about 10 MHz. So thinking like "with PCM we upsample only up to 705.6k but with PCM to DSD up to DSD256" does not consider the 2nd oversampling stage which happens in DSD chip in order to reach delta sigma modulator operating rate.

 

So the difference does not lie in higher sample rate. Both cases 1) and 2) from my previous post contain oversampling up to modulator rate, which is somewhere in MHz range. The difference then lies in digital processing quality. All what DAC chip performs before the D/A conversion stage is digital processing, so output numbers are computed from input numbers. I don't know why it is generally so hardly understandable for people that quality of digital processing can make the change. It is like when you process a picture with higher or lower quality algorithms. You get different results.

[bogi]

Correction: which happens in DSD chip -> which happens in DAC chip

[bogi]

5 hours ago, STC said:

I think I didn’t make myself clear. let me try again. 

 

It is like with a small object on a bitmap image. IMO it is about SNR and dynamic range what affects level of object visibility/audibility and its localization precision also in audio case.

In the case 2) with higher quality digital processing before digital signal enters the D/A conversion stage you get better SNR and higher dynamic range than in case 1). I am referring to my 1st post of this thread.

 

Then, difference in quality of digital processing between 1) and 2) results also in different level of content distortion (which is present in every digitally processed material, be it bitmap image or digital audio content).

[bogi]

1 hour 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.

 

There is no other difference between cases 1) and 2) than the difference in digital processing. If that difference would not exist, I would not use HQPlayer case 2). And many others with me. The case 2) is more complicated to set up by user than 1) since it requires to learn and evaluate new things and it also requires high performance computer preferably with modern nVidia card. Why would so many people use such a solution if it wouldn't have an audible effect?

I am sincerely attempting to understand things and not being a professional helps me easier to trust my ears and accept that my listening experience need not always fit to all information coming from pro world. Despite not being a pro I see that for example Amirs measurements are done too quickly and are much restricted. He is not interested to do time domain measurements except of evaluating SINAD number. He is not interested to measure above 20kHz. He ignores possible intermodulation distortion impact. He does not recognize noise impact coming from computer and power supplies to computer connected analog devices because he is not able to measure the impact which so many people are able to hear. He ignores DSD domain measurements. He ignores software upsampling topic at all. There are too many people who can distinguish 2 DACs with high SINAD numbers, impact of different power supplies, effects of USB noise treatment etc. How easy it is to simplify things so much that too simplified interpretation of restricted measurements contradicts to listening experience of so many people. From my point of view of a non pro person it creates rather comical image of some persons in a pro world. Then, of course, I evaluate which people provide information which better correlates with my hearing.

 

I only guess that many of the audible differences which we can hear but are not measured have base in time domain. I doubt it is enough well discovered what we are able to distinguish in time domain area and I also think that differences in hearing abilities between people are generally not enough considered.

[bogi]

1 hour ago, STC said:

An example is when you hear a ticking clock in very quiet room and the same room with the air cond noise. You still will able to localize the clock because the brain decode the timing difference between the ears of the arrival of the original sound. 

 

When background sound is not far from white noise then yes, you have chance to hear low level clock ticks and localize the source. It gets much harder with for example quickly changing sounds of pop music content as low level background. And if we could add a distortion to the clock sound, which would make the tick transient more difficult to distinguish, I guess it would affect our localization ability too.

 

Looking at some measurements I see that background noise profile often has many spikes, so it doesn't have character of white noise. Other aspect is that measurements are done under very specific conditions (measurement device is calibrated, only DAC output is measured). That generally doesn't reflect real situation with complete audio chain. I have seen for example measurements before applying Intona or HS02 isolator which showed many spikes in much higher noise floor with than standard DAC measurements were showing. Although such a ground loop may be an extreme, I think it is correct to say that pointing to high SINAD number measured under artificial conditions does not reflect specifics of real listening conditions.

 

I think that nothing else than noise floor character and level, SNR and signal distortion is responsible for more flat sound with worse imaging.

[bogi]

Of course, Miska will not explain how he does source content analysis and the details of filter/modulator implementation.

I really don't expect any soundstage specific processing in HQPlayer. I also don't expect a simple filter or modularor parameter affecting soundstage in some specially significant extent. IMO better soundstage simply comes out of more accurate reconstruction of analog signal from source digital content.

 

Soundstage is created in our brains. I agree with the content @fas42 posted above. If the level of signal reconstruction accuracy is enough for our brain to allow to create better image of the scene, our subconsciousness will construct it automatically. I think quite generally, not only in audio area, we are underestimating the function of subconsciousness, all the automation it does for us to live easier. I support the idea fas42 brought into this topic about 'filling the gaps' by our subconsciousness.

[bogi]

On 10/14/2023 at 8:39 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.

I personally prefer PCM to DSD upsampling already about 9 years. I started to use it initially with foobar2000, since it made sound more natural for my ears. With HQPlayer I perceive improved sound quality and slightly different sound with every filter or modulator. Both sound quality improvement and the ability to tune sound according to my personal preferences are the main points for me why I am using HQPlayer.
 

On 10/14/2023 at 8:39 PM, Archimago said:

I'm wondering though, has there ever been a documented listening test with controls in place (eg. same output level) showing that there has been a change in soundstage for the "better" as assessed by multiple listeners, between straight PCM playback and the same data upsampled to DSD256? With which DAC? With what upsampling software and settings?

Like I mentioned above in some post, soundstage is created in our brains. We are clearly coming into subjective waters here. Then we need to assume a listener to be part of a listening chain and attribute listening test results to a specific listener. IMO no generalization of individual listening test results is appropriate. I also cannot associate such a listening test results to word 'objective'.

It is nice if a group of listeners agrees on some commonly perceived listening impressions. But they are no more true than impressions of other persons whose listening experience is different. Listener group selection can also affect results.

 

Over the years I found so much demands on double blind listening tests in endless objective vs subjective discussions at many forums. But those discussions are mostly abstracting from what is happening in listeners head. It may be interesting to read about somebody else's experience, but then I take it only as an interesting reading and not as something what is or should be relevant to me as a person and to my listening chain. We are simply different persons with different hearing abilities, taste and preferences about what is good sound. And we are using different devices, software and their settings. Too much variables and too many individual preferences.

 

There are many aspects of sound quality, one of them is soundstage. People can easily disagree on anything sound quality related, including which aspects of sound quality are more important etc. Hunting for objectiveness does not bring sense to me.

[bogi]

5 hours ago, fas42 said:

Good sound quality delivers everything. Including soundstage. The biggest problem is the general poor quality of measurement methods, which are in the league of the Blind Men and an Elephant parable - small parts of the story can be determined with great accuracy, but the gestalt, which is so obvious at a subjective level, is MIA ...

 

If there would be some software available which analyses measurement results and gives something like "soundstage quality index", then such a measurement results could be called "objective" - given we know and understand how the index was computed. The result would more or less reflect what most of people is perceiving, depending on the analysis method, but at least it would provide the same result in repeated measurements.

 

AFAIK we are very far from having a trustworthy software available giving us some higher level sound quality information.

 

As soon as we substitute measurement gear with group of people, then we are "measuring" opinions of those people and not the listening chain playing some audio content  - since the thing which changes are people listening to the same audio content using the same listening chain. It's like repeating measurements with different measurement gear of totally random quality including buggy ones, calibrated and not calibrated ones etc. and then attempting to "evaluate" soundstage relevant outcome of such repeated measurements in some way.