HQ Player

[giordy60]

8 hours ago, The Computer Audiophile said:

Did you update the Ares firmware as well?

 

Denafrips has firmware that limits units to 16fs and firmware that enables 32fs. 

hi Chris
where did you see the firmware that enables 32fs on Ares 2?

on the denafrips site I have not seen anything .....
escaped me? 

[Miska]

41 minutes ago, giordy60 said:

where did you see the firmware that enables 32fs on Ares 2?

on the denafrips site I have not seen anything .....
escaped me? 

 

If I recall correctly, it is bundled with the other firmware version. The one step older is the one that has 32x PCM and DSD1024.

 

[The Computer Audiophile]

I'm not totally sure which Denafrips products this work on, but my guess is it work on all the USB devices because they should be the same. However, use at your own risk, or please correct me if I'm incorrect. I don't want to put out misinformation. 

 

 

https://www.denafrips.com/firmware-update-usb

 

 

FIRMWARE LOG

 

V3.3.2
- Original firmware

V3.3.5
- Increase USB resync time to address USB out-of-sync issue

- Supports PCM1536 / DSD1024

V3.4.1
- Fixed compatibility issue with macOS  

- PCM1536 / DSD1024 supports disabled

[ted_b]

Interesting that the Holo firmware timeline is similar, in that only the penultimate firmware is capable of 32fs/DSD1024.  The latest one disables that.

[The Computer Audiophile]

Hi @Miska I just noticed HQP supports Merging products such as the Anubis. I have an Anubis Premium. How does HQP work with these products?

 

Edit: Ah, I think I get it now. Just send audio to the virtual Ravenna driver. I wish the products had NAA built in. 

[giordy60]

thanks Chris 👌

[jimdukey]

How disappointing,

they giveth and they taketh away!

[GoldenOne]

Is it possible to disable this popup warning @Miska?

 

It comes up every time I start HQP or change settings. I keep DirectSDM on so that DSD files are passed through, and keep volume on PCM at -3dB to prevent clipping. But this results in this message popping up constantly.

Is there a way to make this no longer show?

[Chodi]

I appreciate the comments about the firmware "upgrade" for the Ares II but that is not the issue here. I finally reverted to HQPlayer version 4.10.3 and all problems with the pcm max rate were gone. Version 4.10.3 does not have any problem with playing pcm at 1.4112M in Linux Mint. This is obviously a bug in the later version of HQPlayer. Unfortunately some of the newer filters are not available in the earlier version but I think I will wait for a future revision before upgrading HQPlayer and hope this problem is resolved. I realize it will only be an issue for a few since most dacs can't reproduce pcm at those rates, but it is something I use often on my Ares II. I think the firmware "upgrade" will wait until I have much more information as I have no problems with the current firmware.

 

I am a little surprised that Jussi hasn't commented on this issue. I guess there are so many things to sort out that this is not a high priority.

[Miska]

8 hours ago, GoldenOne said:

Is it possible to disable this popup warning @Miska?

 

It comes up every time I start HQP or change settings. I keep DirectSDM on so that DSD files are passed through, and keep volume on PCM at -3dB to prevent clipping. But this results in this message popping up constantly.

Is there a way to make this no longer show?

 

DirectSDM disables volume control and sets PCM to fixed value that is slightly higher than  -3 dBFS (3.01... dB) which causes this to appear if volume knob was lower than that at previous HQPlayer close. The last value should get stored at exit, so that you don't get this message on next startup.

 

If you don't limit your volume knob to max -3 dBFS then it shouldn't reappear. (I'm using -6 to 0 dBFS range setting)

 

[GoldenOne]

5 hours ago, Miska said:

 

DirectSDM disables volume control and sets PCM to fixed value that is slightly higher than  -3 dBFS (3.01... dB) which causes this to appear if volume knob was lower than that at previous HQPlayer close. The last value should get stored at exit, so that you don't get this message on next startup.

 

If you don't limit your volume knob to max -3 dBFS then it shouldn't reappear. (I'm using -6 to 0 dBFS range setting)

 

Ah ok thank you.
I'll try removing the volume limiter. Normally keep it there just to prevent it going into potential clipping levels.

Thanks!

[copy_of_a]

@Miska

2 short questions about your "iir plugin" in the matrix processing section:

 

1.) to get a 6db/octave slope for Low- & HiShelves what's the bandwidth ("Q", "S" respectively) I have to set?

Is it S=0.71 or S=0.5?

 

2.) if I use convolution based on Impulse Responses stored in wav-files how do I combine it with additional "plugin" corrections for stereo playback?

Do I enter the iir-plugin values as comma-separated items in the same chanel as the wav files or do I stack two additional source chanels for chanel 1+2

(so...:

slot1: source chanel 1: wav

slot2: source chanel 2: wav

slot3: source chanel 1: iir-plugin

slot4: source chanel 2: iir plugin)

?

 

(I am asking because I want to combine room correction (IRs stored in wav) with different tilt filters that I want to built from cascaded 1st order shelf filters)

 

Many thanks in advance!

 

 

[Miska]

On 5/30/2021 at 11:30 PM, copy_of_a said:

@Miska

2 short questions about your "iir plugin" in the matrix processing section:

 

1.) to get a 6db/octave slope for Low- & HiShelves what's the bandwidth ("Q", "S" respectively) I have to set?

Is it S=0.71 or S=0.5?

 

S=0.5 should do it.

 

On 5/30/2021 at 11:30 PM, copy_of_a said:

2.) if I use convolution based on Impulse Responses stored in wav-files how do I combine it with additional "plugin" corrections for stereo playback?

Do I enter the iir-plugin values as comma-separated items in the same chanel as the wav files or do I stack two additional source chanels for chanel 1+2

 

You just queue all on the same pipeline. So you add convolution file(s) and plugins on the same line, comma separated.

 

On 5/30/2021 at 11:30 PM, copy_of_a said:

(so...:

slot1: source chanel 1: wav

slot2: source chanel 2: wav

slot3: source chanel 1: iir-plugin

slot4: source chanel 2: iir plugin)

?

 

 

This wouldn't give expected result, but instead strange reverb. Because you'd have two pipelines in parallel but with different processing and then mixed together.

 

[copy_of_a]

9 hours ago, Miska said:

S=0.5 should do it.

 

You just queue all on the same pipeline. So you add convolution file(s) and plugins on the same line, comma separated.

 

This wouldn't give expected result, but instead strange reverb. Because you'd have two pipelines in parallel but with different processing and then mixed together.

 

great - many thanks!

[NoNaim]

Really enjoying HQPlayer over optical network after being an MScaler user for a year... normally use Closed-Form-M with NS9. I've tried a mix of other filters but kept coming back to it. Until now. Your new Sinc-Mx with both NS9 and LNS15 is really very, very good!

 

I've tried the PGGB system and while it is good, HQPlayer for me is more useful as a main system given it's ability to use Qobuz etc through Roon to HQP NAA on an OpticalRendu. Anyway, just wanted to say great thanks for the great work and looking forward to how HQP develops! 

[jimdukey]

For some reason I no longer get Posts here in my email.

I had that for years, now it's not working.

How I can I fix this?

TIA!

[jimdukey]

I didn't even receive an email with MY Post!

I have Follow Topic selected.

[fredg_31]

Hi Jussi,

 

I really love poly-sinc-gauss filter, on my setup PCM/NS9/192, voices are much more natural than with ext2 or ext3 and I didn't lose any details!

 

Thanks for it, this filter is a top choice for me!

[Chodi]

I owe an apology to Jussi. My problem with HQPlayer and my Ares II dac. It was not at all a problem with HQPlayer. In linux Mint it seems that there were 2 usb audio profiles for Denafrips dac's. One was submitted by Thesycon (the correct one) and one by someone else. Linux Mint defaulted to the wrong audio profile, not the one provided by Thesycon. Since I am new to Linux I had no idea that there could be multiple audio profiles provided for the same dac. After much digging I found the correct Thesycon profile and made it the default. That fixed all my problems and I now have all the rates available just as in the Windows driver version. Should anyone else with a Denafrips dac experience this in Linux you can pm me and I will do what I can to explain how I found the correct profile. The truth is, probably most with some experience using linux would have found this long before I did. Sorry I jumped to blaming HQPlayer for the problem. 

[Jacky820]

On 4/1/2021 at 4:37 AM, lpost said:

I'm sure this is one of the many reasons Jussi does not like Windows. Sorry I can't help, I run HQPe OS to avoid issues like this. It just works.

May I know what CPU ur using now?
I am using I7 8700 and would like to upgrade to AMD 5950x. Just worry the embedded OS is not working in AMD CPU.

[lpost]

7 hours ago, Jacky820 said:

May I know what CPU ur using now?
I am using I7 8700 and would like to upgrade to AMD 5950x. Just worry the embedded OS is not working in AMD CPU.

i7 6770.

[Yviena]

Hmm idk what you did miska but I can do Ext-3 EC7 DSD256 without drops-outs on windows now with grayed = auto DSP and CUDA unticked with around 9-12% cpu usage.

[Jacky820]

9 hours ago, Yviena said:

Hmm idk what you did miska but I can do Ext-3 EC7 DSD256 without drops-outs on windows now with grayed = auto DSP and CUDA unticked with around 9-12% cpu usage.

Good. May I know what display is using? 

[StreamFidelity]

On 6/6/2021 at 10:09 PM, fredg_31 said:

I really love poly-sinc-gauss filter

 

Me too! 👍

 

In image processing, Gaussian filters are used to smooth or soften the image content. In fact, I get the impression that the music is smoother. Even so, the instrument separation and resolution is excellent. The bass is wonderfully accentuated.

 

I like it best for 44.1/48 poly-sincgauss-xla:
Apodizing extra long Gaussian polyphase sinc filter with extremely high attenuation. Optimal time frequency response. For SDM outputs, processing is two stages with 16x intermediate rate.

 

The computing power is quite high for the DSD256 / ASDM7EC, but still in the green area.

 

The same filter without apodization poly-sincgauss-xl for 88.2/96 pushes my linear power supply with 2 Keces P8 to the limit. I therefore use poly-sincgauss-long:
Long Gaussian polyphase sinc filter with extremely high attenuation. Optimal time-frequency response. For SDM outputs, processing is two stages with 16x intermediate rate.

 

 

[sledwards]

1 hour ago, StreamFidelity said:

In image processing, Gaussian filters are used to smooth or soften the image content. In fact, I get the impression that the music is smoother. Even so, the instrument separation and resolution is excellent. The bass is wonderfully accentuated.

I definitely agree with your analogy to image processing. On my May KTE DAC, I can't run the poly-sinc-gauss-xl or xla when upsampling to 1.4/1.53 Mhz PCM but the long version works nicely. If you have not tried his other new filter; poly-sinc-xtr-short, linear or minimum phase, they both sound fantastic as well. I actually prefer them with my setup over the Gaussian filter family.