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HQplayer - do I need it?

HQP cannot and does not improve anything! It's a software intended for the audiophile believers so they redeem themselves and feel better for committing the sin of using digital music, even if they already have an audiophile-grade linear PSU inside the PC...
I don’t believe I have ever seen any unsighted/level matched comparisons which prove a difference in audibility, like so much of Hi-Fi it is belief based.
Keith
 
I don’t believe I have ever seen any unsighted/level matched comparisons which prove a difference in audibility, like so much of Hi-Fi it is belief based.
Keith
I don't believe you ever read and properly understood this post?

 
Yes I have, what percentage of HQ players purchasers use it for that purpose?
For most it is precisely a ‘M-scaler’ thing, ie no audible difference .
Keith
 
Yes I have, what percentage of HQ players purchasers use it for that purpose?
Quite a few people on other forums

Any talk of percentage and you and I would be talking out of our bums - I don't have time for that.

Hopefully from my linked post you can agree it is a useful tool when used in such a way ?
 
Yes I agree ideal for your purpose, I am afraid for me it will always be associated with ‘computer audiophile’s ’ subjective nonsense, linear power supplies, stripped down OS expensive audio computers , latterly boutique ethernet switches etc etc etc.
Which I have to admit I got totally caught up in, until I tried every suggested tweet for myself.
Keith
 
Yes I have, what percentage of HQ players purchasers use it for that purpose?
For most it is precisely a ‘M-scaler’ thing, ie no audible difference .
Keith
Just my 2 cents, but I'm looking for a software solution where I can input all my music sources (local music files, streaming via Wiim mini) and use DSP / room correction on them. HQPlayer looks like a possible solution - I think there are probably quite a few users out there using it for that purpose. Although, if I also need an RME device to input SPDIF, then I think it may be easier and cheaper to just get a MiniDSP device.
 
No, as long as you have bit perfect transfer, any "jitter" from the SPDIF source is completely irrelevant for the ADI-2

The only relevant jitter is at the point of D to A (and A to D , for that application)

That's a quote from RME's @MC_RME

I've tested and have bit perfect transfer , so clocking is a non issue.

Measuring my DAC for jitter at its analogue output using Cosmos ADC, there is no jitter to worry about, with any of its sources:

Did I say anything about jitter? You have 2 clocks in your setup, one is the spdif source the other is the usb async interface. They are always a tiny bit different and sooner or later the buffer in between will under/overflow. HQP has a resync to deal with this situation but it’s also unlikely because it’s using rather large buffers.
 
Did I say anything about jitter? You have 2 clocks in your setup, one is the spdif source the other is the usb async interface. They are always a tiny bit different and sooner or later the buffer in between will under/overflow. HQP has a resync to deal with this situation but it’s also unlikely because it’s using rather large buffers.
The data clock is recovered by any SPDIF receiver from the incoming signal. The recovered clock is used to pace sample playback; Consequently, the buffers will never under/overflow in a properly engineered receiver. The RME products (any many other well-engineered products) employ a PLL to reduce any jitter on the incoming recovered clock. The jitter-reduced clock remains synchronous to the original, but exhibits lower phase noise. Regardless, buffer over/underflow is not an issue.
 
Just my 2 cents, but I'm looking for a software solution where I can input all my music sources (local music files, streaming via Wiim mini) and use DSP / room correction on them. HQPlayer looks like a possible solution - I think there are probably quite a few users out there using it for that purpose. Although, if I also need an RME device to input SPDIF, then I think it may be easier and cheaper to just get a MiniDSP device.
You might also want to take a look at Roon (https://roonlabs.com/) if you have not already, it includes a rich set of DSP functionality.
 
The data clock is recovered by any SPDIF receiver from the incoming signal. The recovered clock is used to pace sample playback; Consequently, the buffers will never under/overflow in a properly engineered receiver. The RME products (any many other well-engineered products) employ a PLL to reduce any jitter on the incoming recovered clock. The jitter-reduced clock remains synchronous to the original, but exhibits lower phase noise. Regardless, buffer over/underflow is not an issue.

Can you elaborate why you think it’s not an issue?
 
Just my 2 cents, but I'm looking for a software solution where I can input all my music sources (local music files, streaming via Wiim mini) and use DSP / room correction on them. HQPlayer looks like a possible solution - I think there are probably quite a few users out there using it for that purpose. Although, if I also need an RME device to input SPDIF, then I think it may be easier and cheaper to just get a MiniDSP device.

JRiver will do all that. It has a WDM input, so you can use it to play ANY sound on Windows through JRiver. This includes system bleeps (which you will want to turn off!), Youtube, ALL streaming services, and in fact anything that makes any sound at all. It can also convert WDM into ASIO. The downside is: these streaming services are not integrated into the JRiver interface, the same way that Tidal is integrated into Roon. If you want to play music from, say, Idagio ... you have to use Idagio's native app or webpage to play music.

To enable this, you need to do two things:

1. In JRiver, go to Tools - Options - Audio - Advanced. Scroll down and look for WDM driver, then make sure the check mark next to it is ticked.
2. In Windows, go to Settings - Sound - Choose where to play sound. Choose JRiver as the output.

I am currently using JRiver as my "one size fits all" toolbox. Its main use is as a file player for locally stored FLAC files on the HDD, which it convolves with filters and generates an 8 channel crossover, which is then sent to the DAC.

AFAIK HQPlayer can not take WDM input (despite me begging Jussi many times over the years for him to enable this on the other forum), so it is not suitable. I would be extremely happy to be proven wrong though.
 
Yeah, there’s little reason to use HQP over JRiver apart from access to many filters. My impression was that Jussi is refusing to add WDM or ASIO input support to the desktop version because he wants to get additional revenue from sales of embedded licenses, which is not to say there’s anything unusual about this approach.
 
Yeah, there’s little reason to use HQP over JRiver apart from access to many filters. My impression was that Jussi is refusing to add WDM or ASIO input support to the desktop version because he wants to get additional revenue from sales of embedded licenses, which is not to say there’s anything unusual about this approach.

The embedded version of HQP is able to do this? Accept WDM and ASIO input?

That's my other problem with HQP, there is no manual, the webpage has very little information, and the only source of information is that 800 page thread (I forget how many) on the other forum. It's basically impossible to trawl through.
 
The embedded version of HQP is able to do this? Accept WDM and ASIO input?

That's my other problem with HQP, there is no manual, the webpage has very little information, and the only source of information is that 800 page thread (I forget how many) on the other forum. It's basically impossible to trawl through.
HQP Embedded is Linux only - WDM and ASIO are Windows, so would only apply to HQP desktop.
 
there is no manual
1) Install HQPlayer and you will find the manual
or
2) Download Windows setup executable, open it in 7zip and extract manual directly without installation
the webpage has very little information
The web page is full of information but IMO in too dense form. It is easier understandable for professionals than for usual music consumers. I feel it could be better explained what is what - difference between Desktop and Embedded versions, what is NAA, what are typical use cases. The manual is not directly accessible online and more advanced features like matrix pipeline and parametric equalization are not so easily understandable for ordinary music customers. I understand that for people who never tried HQPlayer it is not easy to find orientation in possible usage scenarios and settings.
and the only source of information is that 800 page thread (I forget how many) on the other forum. It's basically impossible to trawl through.
Actually 1101. The basic and accurate information appears in the manual. New HQPlayer users are just asking in that long thread without reading all of those 1100 pages and they are becoming answers. Other place of wide HQPlayer discussion is Roon forum where the author is also responding.
That's my other problem with HQP
If you really want to try HQPlayer, then read the manual first, run trial to get it working at least with default settings and after that post your questions in that long thread.
If you don't have interest in reading the manual and trialing then this forum may be the best place to "learn" about HQPlayer how it is useless. Just start with top of this page.
 
Can you elaborate why you think it’s not an issue?
No audible dropouts

No measurable jitter

How would this "issue" manifest in your opinion?

In practical terms we are talking about literally a non-issue
 
JRiver will do all that. It has a WDM input, so you can use it to play ANY sound on Windows through JRiver.
Does correct sample rate switch automatically? Or manually?

I don't want Windows in any of my daily listening

I boot headless HQPlayer . By the time i sit on my couch I can get music going with just my iPad

User experience is important to some people
 
Can you elaborate why you think it’s not an issue?
Buffer under/overflow isn't an issue when using the SPDIF interface for the reason cited in my earlier post. Underflow/overflow isn't an issue when streaming to a USB audio interface because its internal sample clock is used to pace the DAC samples in that use case. The host need only provide data via USB sufficiently fast to keep the audio interface's output buffers partially full (never empty, nor full) during playback. Unless the output buffers are set to a pathologically-small size (e.g. to mitigate latency), even a 12 Mbps USB2 interface provides ample bandwidth for contiguous audio renders.
 
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