Adventures in Upsampling to DSD

[Ron Scubadiver]

First off the hardware I have available right now is an Intel NUC with an N3700 CPU and a Grace M9xx.  An N3700 has about 33% of the processing capability of a gen 7 i5 two core notebook CPU.  The Grace M9xx can do DSD 256 with the latest firmware upgrade.

 

With Jriver (on trial) the best I could do was DSD 64.  Going to DSD 128 caused skipping.  Next I tried foobar using the SACD input component and it's included DSD processor.  I can get DSD 256 without skipping.  CPU usage with SDM type B is 50%, but the system runs at normal temperatures without the fan ramping up.  That tiny fan can turn 8,000 rpm and if it gets there, you know it.

 

See this link for details on foo SACD input and SDM types:

 

https://diyaudioheaven.wordpress.com/digital/pc-software/foobar-2000-for-dummies-part-3-new-experimental-sacd-plugin-v-0-9-x/

 

SQ is better than no upsampling.  I am less certain if SQ is better than sending 352800/384000 PCM to the DAC upsampled with two instances of Sox Mod 2 set at their defaults.  CPU usage is minimal for PCM output. 

 

Here is the part that I have trouble understanding.  Upsampling to high PCM rates is supposed to bypass upsampling and filtering by the DAC.  The argument goes a computer can do this better with software because it has more resources.  Upsampling to DSD 64 bypasses even more of the DAC which normally turns PCM into something like DSD.  DSD 128 should provide the  additional benefit of moving ultrasonic noise further away from the audio band.  I don't have any idea why DSD 256 would be better or if I had compatible hardware DSD 512.

 

Someone around here noted that some recording studio said they were getting better results with a DSD 255 recording machine, even on child copies at 24/96 and redbook.  That's well and good for recording and down sampling because the process starts with more data.  However, upsampling redbook creates no new data.  It just sends more interpolated data to the DAC.  It would seem all that would accomplish is making all the hardware work harder.

 

There are other issues.  the DSD conversion algorithms are from Phillips.  They are obviously at least twice as efficient as whatever Jriver is using.  I suspect these routines use much less processing horsepower than HQPlayer judging from the relatively powerful computer hardware that group of audiophiles is using to reach DSD 512.  An i5-7600 has around 6 times the processing power of my N3700.  Perhaps HQPlayer and the DSD processing hardware of high end DAC's may have some secret sauce.

[Dr Tone]

2 hours ago, Ron Scubadiver said:

Here is the part that I have trouble understanding.  Upsampling to high PCM rates is supposed to bypass upsampling and filtering by the DAC.  The argument goes a computer can do this better with software because it has more resources.

 

Yes this is true.  With the excellent upsampling/filters/modulators in HQPlayer you can get great results.  With JRiver not so much.  I wouldn't even bother turning it on is JRiver.

[Ron Scubadiver]

15 hours ago, Dr Tone said:

 

Yes this is true.  With the excellent upsampling/filters/modulators in HQPlayer you can get great results.  With JRiver not so much.  I wouldn't even bother turning it on is JRiver.

OK, assuming one is using HQPlaayer, why would DSD 256 or DSD 512 sound better than DSD 128?  Is it just more is better?

[Dr Tone]

44 minutes ago, Ron Scubadiver said:

OK, assuming one is using HQPlaayer, why would DSD 256 or DSD 512 sound better than DSD 128?  Is it just more is better?

 

@Miskais better to answer this question.

[Ron Scubadiver]

I have found an objective reason to use DSD 256 instead of DSD 128 on my foobar based setup.  With DSD 128 there is a double click when starting playback.  That changes to a single click of lower amplitude with DSD 256.  My understanding is this behavior is common, although some combinations of DAC's and players probably can avoid it.

[HardrockInMiniMac]

On 1.1.2018 at 1:18 AM, Dr Tone said:

 

Yes this is true.  With the excellent upsampling/filters/modulators in HQPlayer you can get great results.  With JRiver not so much.  I wouldn't even bother turning it on is JRiver.

 

On 31.12.2017 at 10:57 PM, Ron Scubadiver said:

Here is the part that I have trouble understanding.  Upsampling to high PCM rates is supposed to bypass upsampling and filtering by the DAC.  The argument goes a computer can do this better with software because it has more resources.  Upsampling to DSD 64 bypasses even more of the DAC which normally turns PCM into something like DSD.  DSD 128 should provide the  additional benefit of moving ultrasonic noise further away from the audio band.  I don't have any idea why DSD 256 would be better or if I had compatible hardware DSD 512.

 

You are very close to the answer already. Making it short and giving you the direction to search for an answer by the following example:

 

My Lyngdorf is a PowerDAC (DAC+fully digital amp). It resamples all input signals (even analoge) to 24/96 and continues processing them this way. Most people agree that 99% of all DACs do something similar. So if I upsample 44.1 to 192khz my Lyngdorf will resample that to 24/96 every time. Pointless but it works.

 

So you really got to ask yourself: Where and how is your requested optimization going to take place?

 

The point is to apply software filters while upsampling and people believe that these are superior. Let's think this thru:

 

My DAC has filters embedded so assuming I go upsampling with HQP I can turn all of that off in my Lyngdorf because there is no point in applying more filters to already "filtered" music.

 

Following this I would be left with the room perfect (room correction) features of my Amp but for that I can also use software like Dirac Live and that is also working, I had very good results before I purchased the Lyngdorf.

 

Well, that would be "game over" for Lyngdorf and Tinnov and the likes because any standard Amp can do from that point on and there will be better bang for the buck available. So why did I still buy the Lyngdorf? Vinyl.

 

The reason why I decided for a hardware solution is that I was deeply impressed with DIRAC Live and the effect of room correction more than with any other tweak before and there is absolutely no reason and no compelling way to feed a 5k $ analog chain into a computer in order to have its output run through software room correction and then finally into the Amp. That's a no no no go in the analoge world.

 

The only way is to find an audiophile hardware DSP with a highend analogue input and there are less than a handful such Amps in this market niche (Lyngdorf, Tinnov, McIntosh...). Vinyl is driving my decision and that surely makes me kinda non-mainstream on this forum.

 

But for you, still the same rules apply: Computer based solutions are more flexible and less vendor related. Highend hardware solutions offer enough resources to perform upsampling on the fly but with less flexibility for the user - on the other hand source, software etc. become less important. 

 

I have to add that Hardware solutions are much more expensive than -say- Mac mini with HQP and Dirac Live feeding a standard DAC feeding a standard Amp (which is what I had before and it was already pretty nice).

 

Hope this helps a bit,

Chris

 

 

[Miska]

4 hours ago, HardrockInMiniMac said:

The reason why I decided for a hardware solution is that I was deeply impressed with DIRAC Live and the effect of room correction more than with any other tweak before and there is absolutely no reason and no compelling way to feed a 5k $ analog chain into a computer in order to have its output run through software room correction and then finally into the Amp. That's a no no no go in the analoge world.

 

The only way is to find an audiophile hardware DSP with a highend analogue input and there are less than a handful such Amps in this market niche (Lyngdorf, Tinnov, McIntosh...). Vinyl is driving my decision and that surely makes me kinda non-mainstream on this forum.

 

You can do digital room correction for analog (and digital) inputs with HQPlayer Embedded. I have analog input running at 768/32 resolution and output at DSD256...

 

[Ron Scubadiver]

Perhaps I lack the desire, or I can't hear that well.  I am on the old side.  With the hardware I have, upsampling in software was no better than choosing a different filter which is possible on the Grace M9xx.  I tried HQPlayer a bit.  It doesn't suit me.  Perhaps there is a way to use it with a difference interface and with some DAC's it really sings, but I don't have time or money for that.

 

[Dr Tone]

17 minutes ago, Ron Scubadiver said:

Perhaps there is a way to use it with a difference interface

 

Roon

[Ron Scubadiver]

After a great deal of experimentation I found Foobar using Sox Mod2 with HQ, minimum phase and 90% bandpass produced the best results.  I couldn't get Jriver to sound right no matter what.  It may have been that I am using a DSP to boost at 40hz.  I must be one of those people who does not like pre-ringing.  Note that the Grace M9xx has 4 filter options and minimum phase slow roll off is one of them.  In the documentation for the AK chip there are specs for the filters going up to 192k, so I believe it is reasonable to believe at 352.8/384 there are no hardware filters.

 

One thing I have noticed is JRiver has a sort of antipathy towards upsampling with their fixed parameters on Sox and repeated statements that they will not implement options.  That seems kind of strange for a playback program with so many options.

 

Foobar's main drawback is it is kind of a kit.  Lots of assembly is required.

[buonassi]

On 1/2/2018 at 9:52 PM, Ron Scubadiver said:

Perhaps I lack the desire, or I can't hear that well.  I am on the old side.  With the hardware I have, upsampling in software was no better than choosing a different filter which is possible on the Grace M9xx.  I tried HQPlayer a bit.  It doesn't suit me.  Perhaps there is a way to use it with a difference interface and with some DAC's it really sings, but I don't have time or money for that.

 

 

To me the benefits of upsampling have been diminished with my new DAC - which also has selectable filters.  So it's not just you.  I've read about the M9XX and almost bought it for this very reason but I went with the iDAC2 by iFi.  As OP stated, if your software can do a better job of filtering and interpolation than the DAC, then it becomes easier to hear the increase in SQ when you bypass its filters.  The main benefit for me, is that I don't care for the lively ringing, harmonic excitement etc of traditional filters, and I'm able to dial in my filter parameters to my liking, controlling total overall ringing with the slope of the rolloff, as well as where that ringing takes place (pre or post transient) with a phase slider.  

 

Another argument for upsampling is that you can turn any OK DAC into a very good sounding DAC provided it has a quality analogue output stage and you know how to set your software upsampling parameters.  Saving some money in the process.

 

A note on SoX - it doesn't have the fine control over it you get with iZotope - at least as it's implemented in Audirvana Plus. So SoX has not been a keeper for me.  It sounds exciting at first, but I grew tired of it pretty quickly.  Fatiguing.  Then again, I like a dry vs a shimmery sound and value attack of transients over separation of instruments.  

 

Anyway, back to DSD upsampling.  I've been playing around with it now that my DAC supports it.  There's something there that I like subjectively.  I would say it makes the sound quicker and sweeter, like others have described as sounding more analogue.  The edges of transients seem to be more natural.  Soundstage depth is also improved.  I'm not sure how it manages this, but there's something there.  Too bad it's quite CPU intensive. 

 

[Ron Scubadiver]

15 hours ago, buonassi said:

 

To me the benefits of upsampling have been diminished with my new DAC - which also has selectable filters.  So it's not just you.  I've read about the M9XX and almost bought it for this very reason but I went with the iDAC2 by iFi.  As OP stated, if your software can do a better job of filtering and interpolation than the DAC, then it becomes easier to hear the increase in SQ when you bypass its filters.  The main benefit for me, is that I don't care for the lively ringing, harmonic excitement etc of traditional filters, and I'm able to dial in my filter parameters to my liking, controlling total overall ringing with the slope of the rolloff, as well as where that ringing takes place (pre or post transient) with a phase slider.  

 

 

It's definitely YMMV.  BTW, the Grace suffers from pops and clicks when playing back DSD, and especially if switching back to PCM.  This is one of the things they don't tell you in the reviews.  

[buonassi]

50 minutes ago, Ron Scubadiver said:

It's definitely YMMV.  BTW, the Grace suffers from pops and clicks when playing back DSD, and especially if switching back to PCM.  This is one of the things they don't tell you in the reviews.  

Wonder if it's a usb cable issue? My idac2 so far has been problem free.  There's the usual clicks when you first start playback or skip a track, but after that, its all smooth and no pops even when the tracks change by themselves. 

[Ron Scubadiver]

4 hours ago, buonassi said:

Wonder if it's a usb cable issue? My idac2 so far has been problem free.  There's the usual clicks when you first start playback or skip a track, but after that, its all smooth and no pops even when the tracks change by themselves. 

I have noticed with foobar upsampling to DSD 256 produces a barely audible click.  If I set the player up so that everything goes to DSD, there are no clicks when reverting to PCM, because it doesn't happen.  I am not sure if SQ is improved.

[Miska]

6 hours ago, Ron Scubadiver said:

I have noticed with foobar upsampling to DSD 256 produces a barely audible click.  If I set the player up so that everything goes to DSD, there are no clicks when reverting to PCM, because it doesn't happen.  I am not sure if SQ is improved.

 

You need to compare different conversion algorithms, because with a delta-sigma DAC the algorithms define large portion of the quality/performance...

 

[Ron Scubadiver]

4 hours ago, Miska said:

 

You need to compare different conversion algorithms, because with a delta-sigma DAC the algorithms define large portion of the quality/performance...

 

Well, Foobar uses something from Phillips called SDM and I used SDM B out of the 4 choices, A, B, C and D.  Chances are it isn't as good as what you are using and my computer is resource constrained having only an N3700 processor.  I am getting the best results with the minimum phase, sharp rolloff setting available on the M9xx and letting the DAC do all the work.  It isn't all that sharp of a rolloff, about 90/110.  This is what Bryston BDA-3 uses and it has the AK chip too.

[Miska]

On 1/19/2018 at 4:09 PM, Ron Scubadiver said:

Well, Foobar uses something from Phillips called SDM and I used SDM B out of the 4 choices, A, B, C and D.  Chances are it isn't as good as what you are using and my computer is resource constrained having only an N3700 processor.  I am getting the best results with the minimum phase, sharp rolloff setting available on the M9xx and letting the DAC do all the work.  It isn't all that sharp of a rolloff, about 90/110.  This is what Bryston BDA-3 uses and it has the AK chip too.

 

With AKM chips it is also important to pay attention to check whether the chip is in "DSD Direct" mode. For example RME ADI-2 and TEAC UD-503/NT-503 have this feature.

 

[Ron Scubadiver]

25 minutes ago, Miska said:

 

With AKM chips it is also important to pay attention to check whether the chip is in "DSD Direct" mode. For example RME ADI-2 and TEAC UD-503/NT-503 have this feature.

 

I don't know if my Grace M9xx has that feature.  If it does, it would have to be by default because there is no option for it on the setup interface.

[Miska]

2 hours ago, Ron Scubadiver said:

I don't know if my Grace M9xx has that feature.  If it does, it would have to be by default because there is no option for it on the setup interface.

 

You could ask Grace how they've programmed DSD mode to work. But since according to their documentation the volume control is digital, then it very likely isn't in DSD Direct mode because that would disable the digital volume control... So very likely the DAC isn't really suited for DSD use but instead could benefit from upsampling in PCM domain and being fed at highest possible PCM rate.

 

[Ron Scubadiver]

38 minutes ago, Miska said:

 

You could ask Grace how they've programmed DSD mode to work. But since according to their documentation the volume control is digital, then it very likely isn't in DSD Direct mode because that would disable the digital volume control... So very likely the DAC isn't really suited for DSD use but instead could benefit from upsampling in PCM domain and being fed at highest possible PCM rate.

 

I think your key statement is "DAC isn't really suited for DSD use..."  I may give upsampling to 384k another shot.  The N3700 has more than enough capability to handle it with SOX.  It will have to wait for tomorrow as I am out for the weekend.

[Tsarnik]

On 20/01/2018 at 6:15 PM, Miska said:

 

You could ask Grace how they've programmed DSD mode to work. But since according to their documentation the volume control is digital, then it very likely isn't in DSD Direct mode because that would disable the digital volume control... So very likely the DAC isn't really suited for DSD use but instead could benefit from upsampling in PCM domain and being fed at highest possible PCM rate.

 

Hello Miska,

 

Wouldn't that also apply to the Mytek Stereo192-DSD DAC as well?

I think you already answered that question when you listed the Stereo-192DSD as a recommended DAC for HQP.  :-)

 

Although my Mytek's 'Volume Control' is normally set to BYPASS I am still using the 'Volume Trim' (set to '-15 dB', corresponding to 3 dB attenuation) to adjust to my amplifier's input stage while also adding some headroom for ISPs (when using the S/PDIF inputs without the AD1896).

 

Back to DSD: The Stereo192-DSD's 'Volume Trim' function is executed by the same volume control "unit" of the ES9016 that also handles the 'Volume Control' function (when set to DIGITAL), isn't it?

The fact that the ES9016 applies a digital LP filter to DSD sources in any case makes the ES9016 chip a chip without a DSD Direct mode (as e.g. compared to the DSD179x family with their analogue FIRs), do I understand this correctly?

So here's my question: Would I, in fact, gain anything (pun intended) if I would DISABLE 'Volume Trim' in the Mytek and instead reduce the 'replay gain' in Foobar by a further 3 dB — i.e. gain anything but 3 dB more shaped noise going out to my amp?

[I am so sorry for not using HQP....yet?  ;-\ ]

 

So far I have been quite happy with the SQ improvement of upsampling Redbook to 8fs with SoX and further to DSD128 with foo_input_sacd (SDM Type D) and my i7-6700HQ even does this silently while on battery. :-)

 

PS: BTW, the correct link the other day should have been: http://www.springer.com/us/book/9789811042881

[barrows]

My understanding with the ESS chips is that they take incoming DSD, and then make it multibit, but they do not down convert the sample rate.  This way they avoid the possibility of creating digital artifacts through the decimation process (good) and retaining the full sample rate of DSD.  They basically make the data stream DSD wide, a 32 bits, so they can then apply their (excellent) digital volume control.

I am sure @Miska will correct me if understand this incorrectly.  So it is not "DSD direct" in the strictest sense, but is also not DSD-PCM conversion.

Anyway, if the Mytek applies their volume trim function via the ESS' digital volume control, there will be no advantage to setting it "off" or not, they probably just have this option in settings to make the user feel good, as the ESS volume control does absolutely nothing to the data when set to full scale, whether it is "on" or "off". 

[MacAudio]

a bit OT.

I also use a Mytek 192DSD DAC at the moment and only use the Mytek digital volume control. Only settings from -20dB up to zero. Volume Trim menu is disabled. Instead i placed the Jumpers to -6dB internally.
Is the digital ESS volume control  better than the often preferred analog version of the Mytek 192 dac ?

[barrows]

1 minute ago, MacAudio said:

a bit OT.

I also use a Mytek 192DSD DAC at the moment and only use the Mytek digital volume control. Only settings from -20dB up to zero.
Is the digital ESS volume control  better than the often preferred analog version of the Mytek 192 dac ?

Yes, if you want the highest resolution and uncolored sound.  As long as you do not need too much attenuation (and -20 dB is nowhere near too much) the ESS VC is transparent.

Really good analog volume controls are very, very expensive, and at the price of these DACs I doubt that they could have a really good analog volume control.

[MacAudio]

Thank you for the fast reply ;-)
it is supposed to be a handpicked PGA2311 from Texas Instruments. For the HP output is another same chip working.