[shwnwllms]

Hey thanks for the quick reply, I tried every option from 512 all the way down to 64 with the same result (no sound comes through). I'm beginning to think maybe my DDC IIS isn't fully compatible with the IIS on the Gustard, but PCM works up to 384, weird.

Not too familiar with the A18 but just looked up the specs. Over I2S looks likes it’s only capable of DSD256, and even then that’s DoP

Switch bit rate limit to 256 x 48, then under SDM pack choose DoP (output tab above where 48K is checked)

 

[Coolmoose]

Not too familiar with the A18 but just looked up the specs. Over I2S looks likes it’s only capable of DSD256, and even then that’s DoP

Switch bit rate limit to 256 x 48, then under SDM pack choose DoP (output tab above where 48K is checked)

Nope, not working. The display on the dac switches to DSD for a brief moment then immediately switches back to PCM, with no sound coming from the headphones. I'll try it with another DDC to see if it solves the issue. Thanks

 

[shwnwllms]

Nope, not working. The display on the dac switches to DSD for a brief moment then immediately switches back to PCM, with no sound coming from the headphones. I'll try it with another DDC to see if it solves the issue. Thanks

Try updating your AISO drivers if you haven’t already. I’m sure someone with an A18 be able to provide better help

 

[bogi]

I recently replaced an office dac with a Gustard A18, running IIS through an Amanero USB bridge, and now upsampling to SDM would not work(no sound) and PCM would only do 384k max, there is also 1 -2 second static/popping noise before the music starts playing.

I would rather add galvanic isolation to USB input instead of using I2S DDC. Topping HS02 or Intona USB isolator. It has positive impact on sound. It may be also cheaper and doesn't require power supply. It's easier to setup and it is invisible for both computer and DAC. It cannot cause DSD rates being not playable etc.

Asynchronous (buffered) USB input will always show better jitter pictures. With USB, DAC clock controls reading from input buffer. With I2S DAC clock is derived from external I2S clock, DAC clock needs to adapt. Clock signal transferred through longer I2S cable is prone to be affected by EMI.

 

[Coolmoose]

I would rather add galvanic isolation to USB input instead of using I2S DDC. Topping HS02 or Intona USB isolator. It has positive impact on sound. It may be also cheaper and doesn't require power supply. It's easier to setup and it is invisible for both computer and DAC. It cannot cause DSD rates being not playable etc.

Asynchronous (buffered) USB input will always show better jitter pictures. With USB, DAC clock controls reading from input buffer. With I2S DAC clock is derived from external I2S clock, DAC clock needs to adapt. Clock signal transferred through longer I2S cable is prone to be affected by EMI.

I'll definitely consider the topping HS02, thanks for the suggestion.

Try updating your AISO drivers if you haven’t already. I’m sure someone with an A18 be able to provide better help

Yea, I tried reinstalling the lastest firmware from Amenero, I even tried the AISO4ALL drivers, but nothing works. Oh well, its not a big deal, i'll try the HS02 or just settle on listening through coax. I was just making sure I didn't make a silly mistake in setting up HQplayer. Thanks for the help.

 

[Louisiana]

This morning is running:

Impressive!

 

[Patu]

This morning is running:



Impressive!

Why do volume leveling in Roon? It's unnecessary and not recommended in combination with HQP. You'll want to feed bitperfect signal to HQP and you can adjust volume level in HQP.

 

[Louisiana]

Why do volume leveling in Roon?

It is necessary for me
I stream 95% of my music, and the differences in playback volume are simply enormous!
I personally hear absolutely NO difference with volume leveling enabled compared to disabled.
HQPlayer can only do volume leveling for local music libraries, but I'd be happy to be proven wrong

 

[Patu]

It is necessary for me
I stream 95% of my music, and the differences in playback volume are simply enormous!
I personally hear absolutely NO difference with volume leveling enabled compared to disabled.
HQPlayer can only do volume leveling for local music libraries, but I'd be happy to be proven wrong

After I wrote the message, I thought that this might be the case. I usually play one album at time and yes, volume levels differ betweem albums, but I've never had problems with it. I simply use the volume control of the preamp/headphone amp. I mostly play local files but also stream with Qobuz.

 

[plumpudding2]

It is necessary for me
I stream 95% of my music, and the differences in playback volume are simply enormous!
I personally hear absolutely NO difference with volume leveling enabled compared to disabled.
HQPlayer can only do volume leveling for local music libraries, but I'd be happy to be proven wrong

Indeed for volume leveling from Qobuz you need to use Roon. I really like the volume leveling feature, if I set my amp at the voltage required for 94dB and Roon volume levels to -14 lufs I can be sure i'm safely listening to 80db K-weighted average volume
I've noticed that especially with rock music 80db is much lower than what i'd consider the music sweet-spot..

 

[Ghoostknight]

does someone know if the "dither" option under "PCM" is used for pcm to dsd upsampling?

 

[roderickvd]

No, it’s not. DSD is inherently noise-shaped as per the modulator order.

 

[wagninger]

Why do volume leveling in Roon? It's unnecessary and not recommended in combination with HQP. You'll want to feed bitperfect signal to HQP and you can adjust volume level in HQP.

It’s not bitperfect then anymore? I thought it’s just using metadata to change gain settings, not lowering the literal volume

 

[roderickvd]

No. It retrieves the appropriate volume gain from the song’s metadata, then alters the samples to the appropriate amplitude.

 

[roderickvd]

I mentioned it already today ... see filter table in HQPlayer PDF document, the last column. Starts on page 25. You can see polynomial filetrs are not apodizing so yes, they preserve original sample values. In the manual you find also filter description - they don't produce perceivable pre- or post-rignging, but their reconstruction quality is low and thus Miska marekd them in teh manual as not recommended. It is worth to open the manual, particulatly the tables of filters or modulators (or dither in PCM case).

I think we are talking about two different things. I understand that apodizing filters can alter samples (for the better) and know that the manual clearly marks which filters are apodizing. Different from apodization, I was referring to this blurb from Schiit:

"The math involved in developing the filter and calculating has a closed form solution. It is not an approximation, as all other filters I have studied (most, if not all of them). Therefore, all of the original samples are output. This could be referred to fairly as bit perfect; what comes in goes out." (emphasis mine) This and more under the FAQ at https://www.schiit.com/products/yggdrasil.

Nearly every other filter -- apodizing or not -- does not share this trait. With those other filters, when you input a sample value x at audio stream position t, and you inspect that sample value t at the output (so awaiting the latency of the filter), the value is almost certainly not x anymore. What's better is another matter, but this in=out thing and only interpolating in-between is something particular to that closed form filter.