[babarrun]

Does popping happen regardless on target sample rate and filter/modulator selection?
Is your target sample rate in the same family (44.1x or 48x) as source content played? If not, try if changing target rate to source rate integer multiple helps.

I can't find the option to change the target rate to source integer multiple.

I am using Tidal and Qobuz hi-res.

any thoughts on what do to in Roon when it comes to MQA? does HQ Player do any MQA unfolding.. the Spring3KTE doesn't as I understand it so was using my DMP-A6 for it earlier. without MQA unfolding doesn't the value of HQ Player decrease or at least would I be better off using Qobuz even if I like it less as a service.

I have attached pictures from Roon and all the options in HQ Player. Hope yo be able to solve for this.

 

[bogi]

I can't find the option to change the target rate to source integer multiple.

[x] Adaptive rate in Output tab. It is tristate check box. You have set the middle (greyed) default setting. It means output rate (limit) you have set in PCM and SDM tabs will be used when playback is possible with given source rate, target rate and filter combination. Actual output rate will be adapted by this setting ONLY if input content is from the other rate family AND the chosen filter allows only integer multiples. With less words: only if necessary. The filter table in HQPlayer manual has a column to see which filters allow only integer multiples (and some only 2^x multiples, so for them the rule applies accordingly).

If this checkbox is fully checked, output rate ALWAYS follows input rate family regardless on filter selected. Integer multiples are easier on processing.

But having it fully checked collides with your [ ] 48k DSD setting in SDM tab. It would work for PCM output but it would make 48k based PCM content unplayable with DSD output. With disabled 48k DSD you tell you cannot play 48k based DSD rates and thus no adaption to 48k based source content is possible.

Advantage of using a DAC which supports 48k based DSD rates is this possibility of target rate adaption according to source rate family.

During playback, you see the actual output rate in the main window.

 

[babarrun]

[x] Adaptive rate in Output tab. It is tristate check box. You have set the middle (greyed) default setting. It means output rate (limit) you have set in PCM and SDM tabs will be used when playback is possible with given source rate, target rate and filter combination. Actual output rate will be adapted by this setting ONLY if input content is from the other rate family AND the chosen filter allows only integer multiples. With less words: only if necessary. The filter table in HQPlayer manual has a column to see which filters allow only integer multiples (and some only 2^x multiples, so for them the rule applies accordingly).

If this checkbox is fully checked, output rate ALWAYS follows input rate family regardless on filter selected. Integer multiples are easier on processing.

But having it fully checked collides with your [ ] 48k DSD setting in SDM tab. It would work for PCM output but it would make 48k based PCM content unplayable with DSD output. With disabled 48k DSD you tell you cannot play 48x DSD rates and thus no adaption to 48k based source content is possible.

Advantage of using a DAC which supports 48k based DSD rates is this possibility of target rate adaption according to source rate family.

During playback, you see the actual output rate in the main window.

thanks. I will almost never play DSD.

I kind of want to set it and leave it so I can freely play music without ever having to open HQ Player agian and get oversampling up to 1.536 which the SpringKTE is capable of.

can you suggest any changes I may need to make to get to my goal? I really appreciate your help on going into rates and filter combinations but that is going over my head.

 

[plumpudding2]

I am not an electronics engineer and not an expert too. But theoretically infinitely long filter with sinc function will recreate the original band widh limited signal back. If longer filters are poorer in time domain then at infinitely long filter should have worst time domain accuracy rather than recreating original signal back. Isn't it?

Due to some quantum limit analogous to the Heisenberg uncertainty principle, a signal can be finite either in time or in the frequency domain but not both! Truly band-limited signals must have infinite length. Because our music is finite in time it must have infinite bandwidth in the frequency domain and that's why the longer filters have poorer time domain performance.

A very interesting paper on the matter is "On bandwith" by D Slepian from 1976 as he examines this apparent paradox.

http://web.eng.ucsd.edu/~massimo/ECE287C/Handouts_files/On-Bandwidth-ProcIEEE.pdf

 

[bogi]

can you suggest any changes I may need to make to get to my goal? I really appreciate your help on going into rates and filter combinations but that is going over my head.

I have no information if Spring KTE is capable of 48k based DSD rates. It is easy to test it - set [x] 48k DSD, set [x] Adaptive rate as fully ticked and set SDM as default mode in Output tab. Then try to play 48k based PCM content. If that is working, leave [x] 48k DSD as is, otherwise uncheck it.

Since you almost never play DSD, I recommend you to have [x] Adaptive rate as fully ticked. That will lower computer resources when you are playing 44.1k based source content (44.1, 88.2, 176.4, 352.8, 705.6, 1411.2). With your current setting these are always upsampled to 1536, which is multiple of 48. With [x] Adaptive rate fully ticked they will be upsampled to 1411.2 with less demand on computer resources.

 

[EMINENT]

Helo.

I have adaptive and 48k checked. Should be same for S3.

 

[Superunknown2]

What is everyone's favorite HQ Player setting for rock/metal?

 

[Superunknown2]

It seems many here prefer using SDM instead of PCM. If you are streaming a FLAC file, can you do SDM? I have been trying to do this, but it always plays as PCM.

 

[genefruit]

It seems many here prefer using SDM instead of PCM. If you are streaming a FLAC file, can you do SDM? I have been trying to do this, but it always plays as PCM.

likely need to change the output screen to SDM rather than PCM or Source.

 

[Superunknown2]

likely need to change the output screen to SDM rather than PCM or Source.

I think I did, but it still plays everything at PCM. Here are my output settings. Should I change something else to play SDM instead of PCM?

 

[genefruit]

I think I did, but it still plays everything at PCM. Here are my output settings. Should I change something else to play SDM instead of PCM?

I'm guessing you need to turn on DoP to pass SDM over PCM

 

[EMINENT]

I think I did, but it still plays everything at PCM. Here are my output settings. Should I change something else to play SDM instead of PCM?

Helo.
If you are using Roon, disable dacs in Audio Devices in Settings.

 

[Superunknown2]

That fixed it, thank you. Do you all also prefer to run music in SDM/DSD rather than PCM?

 

[genefruit]

That fixed it, thank you. Do you all also prefer to run music in SDM/DSD rather than PCM?

I'm typically SDM but results can vary on several factors including personal preferences, compute power, chain, and genre

 

[EMINENT]

That fixed it, thank you. Do you all also prefer to run music in SDM/DSD rather than PCM?

I do.

Also, today I report 3090 install success.
This was delayed from being complete last weekend, I assume due to limited power of V850 PSU which I thought was odd since the 4090 had no issue.
Ordered and installed V1100 now and 14900K OC'd along with 3090 has achieved ASDM7ECV2 hires lp 1024 x 44.1 rates which was previously not possible with 13900K and 4090.