Consensus about upsampling to 512 DSD

[jabbr]

6 hours ago, Audiophile Neuroscience said:

 

Good points Jonathan.

My current DAC does octuple DSD512. It does not however  have a volume control. I could use JRiver for this but that would mean not using ASIO and perhaps degradation of sound for other reasons.

 

I’m not familiar with JRiver DSD upsampling. Typically there is a software setting that does volume, and shouldn’t interfere with the output driver. The way that PCM->DSD conversion works is that there is some type of volume matching, typically a 6dB correction so setting this to something else shouldn’t “degrade” the sound unless, frankly, the software is broken.

[Kal Rubinson]

21 minutes ago, jabbr said:

I’m not familiar with JRiver DSD upsampling. Typically there is a software setting that does volume, and shouldn’t interfere with the output driver.

Unless something has been changed lately, JRiver cannot directly upsample DSD-to-DSDx without an intermediary conversion to PCM.  OTOH, it can (and does) upsample PCM -to- PCMx -to- DSDx.  In both cases, however, it cannot apply variable volume control to DSD output, only PCM.

[audiventory]

2 minutes ago, Kal Rubinson said:

JRiver cannot directly upsample DSD-to-DSDx without an intermediary conversion to PCM

There is a stuff where it described?

[jabbr]

1 minute ago, Kal Rubinson said:

Unless something has been changed lately, JRiver cannot directly upsample DSD-to-DSDx without an intermediary conversion to PCM.  OTOH, it can (and does) upsample PCM -to- PCMx -to- DSDx.  In both cases, however, it cannot apply variable volume control to DSD output, only PCM.

Thanks!

@Audiophile Neuroscience do me a favor and lets try and get your current DSD512 capable DAC to its best performance, I use HQPlayer, but other folks like A+3 -- I'm trialing A+3 now that it does streaming... in the process of trying to get my RPi3B+ recognized so I can stream to "mpd". To use volume control, you need one of these packages that does native realtime upsampling.

[Kal Rubinson]

3 minutes ago, audiventory said:

There is a stuff where it described?

Nope.  Things like that are gleaned from the JRiver YABB forums.    And, btw, my full statement was 

"Unless something has been changed lately, JRiver cannot directly upsample DSD-to-DSDx without an intermediary conversion to PCM. "

[audiventory]

23 minutes ago, Kal Rubinson said:

Nope.  Things like that are gleaned from the JRiver YABB forums.    And, btw, my full statement was 

"Unless something has been changed lately, JRiver cannot directly upsample DSD-to-DSDx without an intermediary conversion to PCM. "

 

There are no known me technology (patents, publications) to alter DSD without PCM intermediate conversion.

But PCM should be considered in wider meaning than 24 bit / 3xx kHz.

 

So it is general matter, not only certain software.

Read details https://samplerateconverter.com/educational/dsd-dsf-dff-editor

[Jud]

13 hours ago, jabbr said:

 

How a “volume control” works on a DSD stream may not be intuitive to you. The DSD stream literally has a component in the digital domain — the carrier at 22-25 MHz — and a component in the analog domain — the audio. The DAC can simply remove the digital carrier and leave the analog signal alone. 

 

The signal hasn’t been “mangled” rather the upsampling is done to a different level. 

 

 

If your amp likes a preamp then by all means use it, rest assured, however, that what the preamp is has doing has nothing to do with volume control, rather amplitude and impedance matching. It will work just as well as when you “volume control” your DSD512 stream — to keep this focused on the topic of this thread...

 

Again, use your preamp along with volume control in your DSD upsampling software! It will work just as well if not better. 

 

BTW: the Bricasti DAC only accepts up to DSD128, so the conclusions you’ve drawn from this example may not apply to the topic at hand. also really hard to know what any one DAC is doing under the hood. 

 

All of these examples will of course be quite specific. Some folks will have good experiences using a preamp, others will prefer what they hear from a digital volume control.

 

I’ve been curious to hear what going direct sounds like, figuring there’s no preamp like no preamp, but three things have held me back:

 

- If I make a mistake with the volume control going direct, my system could possibly play “Pop Go The Speakers.”

 

- I have multiple sources, so going direct with one would involve rearranging wires and hassle.

 

- I am guessing my preamp is of high enough quality that a digital volume control and DAC output section might have to go some to beat it. These days, DACs seem to be developing somewhat faster than other components, so I’m reluctant to spend a lot on one.

[barrows]

11 minutes ago, audiventory said:

 

There are no known me technology (patents, publications) to alter DSD without PCM intermediate conversion.

But PCM should be considered in wider meaning than 24 bit / 3xx kHz.

 

So it is general matter, not only certain software.

Read details https://samplerateconverter.com/educational/dsd-dsf-dff-editor

Yuri:  I am not sure how @Miska does volume control in HQ Player of DSD, perhaps he can clarify for us here.  I use DIY DACs here with ESS 9018 and 9038 chips and my understanding is that these chips control DSD volume something like this:

 

incoming DSD (say DSD 256 so 11.2896/1 bit)-convert to 11.2896/32 bit for volume control-convert to 11.2896/6 bits-convert to analog.  Now some would say this a is a conversion to PCM, but without decimation, so none of the usual artifact problems.  Of course this approach and its sound quality will be dependent on the performance of the DS modulator in these conversions.  Certainly with the power available in computers they could control DSD volume this way, and my understanding is that the only limitation would be the performance of the DS modulator, right?

[barrows]

2 minutes ago, Jud said:

- I am guessing my preamp is of high enough quality that a digital volume control and DAC output section might have to go some to beat it. These days, DACs seem to be developing somewhat faster than other components, so I’m reluctant to spend a lot on one.

Jud, my (former) pre amp was an Ayre K5xe-MP, a quite good unit.  I was surprised at the increase in transparency gained by removing it.  Although I hear your concerns about software based volume controls, one does not want to experience a glitch to full scale even once, as such could clip the amplifier badly and result in speaker damage.  For me, the volume control built into the ESS chips my DACs use has always been entirely glitch free, and I am confident in it.

Are folks using software based volume controls such as HQ Player's 100% confident it never glitches to full scale?

Soon I am going to be trying a DSC-2 type DAC and will need some way to control volume, i have been debating adding an analog VC/gain stage after the DSC-2, or trying HQ Player VC...

[Whitigir]

Audiventory vs Xivero, which one has better DSD upconversion from PCM ? 

[pkane2001]

1 minute ago, barrows said:

Are folks using software based volume controls such as HQ Player's 100% confident it never glitches to full scale?

Soon I am going to be trying a DSC-2 type DAC and will need some way to control volume, i have been debating adding an analog VC/gain stage after the DSC-2, or trying HQ Player VC...

 

I’ve been using HQP volume control directly into the amp from the DAC.  No issues so far, except for one time when I pressed the wrong button in a half-baked USB driver control panel that resulted in a 0dB high pitched sound being sent to the amp. This burned out the right tweeter and hurt my ears. The tweeter was fairly easy to repair as the coil lead wire burned out right at the connection to the  terminal.

 

Obviously this was not a problem caused by HQP, but there is always a potential that some software glitch might cause an unexpected output.

 

 

[audiventory]

47 minutes ago, barrows said:

incoming DSD (say DSD 256 so 11.2896/1 bit)-convert to 11.2896/32 bit for volume control-convert to 11.2896/6 bits-convert to analog.

 

47 minutes ago, barrows said:

Now some would say this a is a conversion to PCM, but without decimation, so none of the usual artifact problems.

 

Artifacts are result of digital filtering, that allow to decimate.

 

Decimated samples consume lesser computing resources.

 

But non-linear processings without the digital filtering can cause audible  noise.

 

Gain altering is linear processing. But non-filetered input DSD's noise can reduce dynamic range of output sigma-delta modulator (convert back to DSD).

To avoid the dynamic range reducing, it is need to use the digital filter, but it cause ringing.

 

There is not ideal decision.

 

In my opinion, ringing is not big matter, because most musical stuff is smooth enough and I know nothing about serious researches of ringing impact to ears.

But broken stability of the sigma-delta modulator at high musical levels due overload is more probable and obviously audible after the dynamic range reducing.

 

 

47 minutes ago, barrows said:

Certainly with the power available in computers they could control DSD volume this way, and my understanding is that the only limitation would be the performance of the DS modulator, right?

 

If there is sigma-delta modulator without filter, then the modulator consume all resources. But filter of DSD noise can consume more resources than the modulator.

[jabbr]

54 minutes ago, barrows said:

Are folks using software based volume controls such as HQ Player's 100% confident it never glitches to full scale?

 

Important to set the volume limits appropriately in HQPlayer, which allows this e.g. never go above -6dB

[jabbr]

1 hour ago, barrows said:

Yuri:  I am not sure how @Miska does volume control in HQ Player of DSD, perhaps he can clarify for us here.  I use DIY DACs here with ESS 9018 and 9038 chips and my understanding is that these chips control DSD volume something like this:

 

incoming DSD (say DSD 256 so 11.2896/1 bit)-convert to 11.2896/32 bit for volume control-convert to 11.2896/6 bits-convert to analog.  Now some would say this a is a conversion to PCM, but without decimation, so none of the usual artifact problems.  Of course this approach and its sound quality will be dependent on the performance of the DS modulator in these conversions.  Certainly with the power available in computers they could control DSD volume this way, and my understanding is that the only limitation would be the performance of the DS modulator, right?

 

Woah! 32 bits at dsd256 is a lot of extra bits which aren't strictly needed for volume control but far be it for me to comment on how the ESS hardware works...

 

Safe to say that HQPlayer works in the single bit SDM domain as far as input/output (although the internals are opaque). As I've indicated, there is a huge digital volume ratio that can be applied before impacting the very best analog electronic SNR, so aside from what anyone decides to implement, the single bit SDM is sufficient from an information content perspective.

[Jud]

1 hour ago, Whitigir said:

Audiventory vs Xivero, which one has better DSD upconversion from PCM ? 

 

I don’t know, but as I recall Archimago said some good things about the Audiventory results. I use Audiventory and like it.

[mansr]

4 minutes ago, Jud said:

I don’t know, but as I recall Archimago said some good things about the Audiventory results. I use Audiventory and like it.

Audiventory applies a very steep lowpass filter near 20 kHz (I don't recall the exact number). Those with a fear of filters should probably avoid it.

[Jud]

42 minutes ago, jabbr said:

 

Important to set the volume limits appropriately in HQPlayer, which allows this e.g. never go above -6dB

 

Yeah, I experienced zero attenuation using HQP with my desktop system a couple of times - no fun. Speakers survived, but I wouldn’t want to try it too often.

 

I did not have the “never go above” setting on.

[Jud]

Just now, mansr said:

Audiventory applies a very steep lowpass filter near 20 kHz (I don't recall the exact number). Those with a fear of filters should probably avoid it.

 

Those with a “fear of filters” would want to avoid digital audio altogether.

 

I don’t know whether Audiventory still offers a minimum phase filter to avoid pre-ringing. I believe there is also a “wideband” option, though you’d have to ask Yuri precisely what it does.

[audiventory]

8 minutes ago, mansr said:

Audiventory applies a very steep lowpass filter near 20 kHz (I don't recall the exact number). Those with a fear of filters should probably avoid it. 

As I said before:

• either fear of filters;
• or lower maximal signal/noise ratio || more probable audible intermodulations || more probable broken stability.

In my opinion, last 2 things are more practically available for perception, than ringing.

 

If we allow to lose N dB, we can allow non-filtered DSD processing to avoid of broken stability.

 

But no one of people who was tried wide band 100 kHz, was stumbled with audible intermodulation (additional noise).

 

When 100 kHz band is applied there is possibility to makes lesser steep filter.

But if we want to maximally keep all information below 100 kHz, there is need steep filter again. Steep filter need to avoid more degradation of dynamic range due DSD noise.

[Ralf11]

9 hours ago, semente said:

 

Pass a signal through an equipment, compare with the original signal. How simple is that?

 

it is too simple

 

re your other question, some distortion products are a given - so, we want the sequence of distortion products to be of minimal psycho-acoustic effect

 

 

agree with your measurements correlate with SQ comment

[jabbr]

2 hours ago, audiventory said:

But if we want to maximally keep all information below 100 kHz, there is need steep filter again. Steep filter need to avoid more degradation of dynamic range due DSD noise.

Could you confirm under which circumstances the 20kHz filter or 100kHz filter are applied.

 

Suppose I have an SACD ISO and which to extract/convert to DSF, is there a filter applied?

[fas42]

10 hours ago, pkane2001 said:

 

So I assume you’ve pulled this off, Frank? Tell us, how you know that what you heard was not all in your head, while the system  was just as pedestrian as all of ours, or as Mr. GUTB would say, it sucked?

 

Yes. Happened nearly 35 years ago, for the first time. Of course it's in my head - the objective quality only improved slightly, by normal metrics; yet the subjective presentation dramatically improved - the Wow! factor to the n'th degree. The most 'radical' aspect was that it became impossible to locate the speaker drivers aurally, even with one's ear only inches from the working surface - this is what it makes trivially easy to assess how close one is to this quality; it's an 'objective' measure.

 

All "pedestrian" rigs are capable of this, but they are kneecapped by sometimes silly weaknesses which add audio anomalies, and the ear/brain registers these - the illusion never materialises. When it first occurred for me, the system would degrade out of this optimal state continually - it was a struggle to get a handle on this behaviour, and I never did so, back then - I abandoned hifi altogether for many years, because the frustration was too much.

[fas42]

10 hours ago, pkane2001 said:

 

Difficult is not the same as impossible. If this can be determined objectively through measurements with proper tools then why insist on using completely unreliable subjective tests, instead?

 

Until "measurements with proper tools" are available, rather than just a thought bubble, then 'subjective' assessment will be required. 'Training' oneself to hear distortion and interference artifacts in the replay adds enormously to one's ability to assess - the downside of this is that it becomes impossible to listen to conventional reproduction for pleasure, because all the deficiencies are now far too obvious.

[fas42]

10 hours ago, jabbr said:

Diagnosis: there is a specific teachable diagnostic method — you have nothing specific. 

 

 

Yes, there is a teachable method - the specific is to use a 'bad' recording to highlight the symptoms - and every time I mention this step, everyone does an ostrich - you know, lead, horse, water, drink ...

[fas42]

9 hours ago, Jud said:

I’ve been curious to hear what going direct sounds like, figuring there’s no preamp like no preamp, but three things have held me back:

 

- If I make a mistake with the volume control going direct, my system could possibly play “Pop Go The Speakers.”

 

 

People have mentioned this a number of times, as a 'danger' - my experience is that drivers are much tougher than given credit for, and a momentary maximum volume blast does zero damage - it's long term, sustained driving of speakers with a badly clipping amplifier that cooks them, or a specialist test track of pure, high level treble.

 

I used to run many of my setups at the maximum volume setting for long periods, for listening pleasure - if there are no audible signs of stress, apart from one's own ears, then no harm done, IME.