HQ Player

[jimdukey]

I thought Direct SDM was just for unprocessed Native DSD, not for any upsampling.

Just got Ver 3.17, just use it for Upsampling to DSD 128/Mac D150.

I love the poly sinc extr mp setting.

Vol set at -3,-3.

Very nice.

I've kept 3.16.4, but don't think I'll need to go back.

On IMac/Sierra, 3.2-3.6 GH Processor.

Some of the set-ups I've read about seem complicated, mine is just the basics-

tho I use the Ifi USB 3.0 suite of products, cost about the same as Micro Rendu.

I don't need things to sound any better than they do now!

Certainly no glitches.

 

[Miska]

19 hours ago, mahi said:

I have some dsd 256 files but I can only play dsd 128 in my current setup. So I will need to look into those dsdiff/dsf settings, which i have not done before.

 

If you uncheck the DirectSDM option, then you can get DSD256 converted to DSD128.

 

19 hours ago, mahi said:

But is it true that the filter and modulator settings in the main interface also apply to dsd resampling as well?

 

Modulator does have effect, filter doesn't...

 

[Miska]

On 8/23/2017 at 1:23 PM, bigbear2003 said:

I am wondering if HQplayer can "mute" the first sec to silent this noise.

 

DAC is supposed to do that, only DAC can control it's output mute relay/transistor. But some DACs have decided to omit such for one reason or another. And that makes the switching noisy...

[bigbear2003]

3 hours ago, Miska said:

 

DAC is supposed to do that, only DAC can control it's output mute relay/transistor. But some DACs have decided to omit such for one reason or another. And that makes the switching noisy...

Is there anything Hqplayer or rpi config. can help to resolve the prob.  Tried Volumio last night and there was no noise.  I guess it is related to the latency PC to rpi to DAC vs. rpi to DAC directly.

[evalon]

Hi all,

 

I hope it is ok that I just briefly jump in on this thread with a question ... (haven't seen it answered doing a search) ...

 

Seeing that computer audio is really evolving I wonder if it is now possible to use HQPlayer to stream music from youtube, Tidal, or some other internet music source? Could e.g. be via Roon, Muso, or Jriver ...

 

Might one of you know if this is now possible - and also reasonably simple to do (I will have a suitably fast PC & DAC but hope I won't need an NAA or the like - so as to keep things simple). 

 

I would much appreciate help in this ;-)

 

Cheers,

 

Jesper 

[k6davis]

On 7/22/2017 at 1:33 PM, firedog said:

I have the same issue. Playlists in HQP/Roon pairing stop at the end of songs.

 

On 8/1/2017 at 1:26 AM, clipper said:

I went back to HQP version 3.15.1.  

 

No problem now.  It will play all day without stopping.

 

On 8/1/2017 at 7:03 PM, Miska said:

 

Yes, seems to happen at the end of DSD track if next track is a PCM track, but not if next track is a DSD track.

 

I'm still having this issue with version 3.17.0. I went back to version 3.15.1 as clipper suggested and like he said, the older version doesn't have the issue at all. Thanks clipper.

 

Unlike what others are experiencing, my system will hang at the end of a track, randomly, regardless of the format. I can put a PCM file on repeat and it will hang at the end after a few plays. The fact that the Roon/HQP combo doesn't have this problem with the older version of HQP suggests to me that the issue is on the HQP side - if it's not something in my configuration, which is of course a possibility. I'd like to stay current with HQP,  but not with the interruptions. 

 

I'd be happy to provide any additional info or logs to help diagnose the problem.

[satbox]

I use HQPlayer version 3.17 with convolution engine enbled (with 2 mono files (L & R) created with Acourate (384 kHz)).

 

In the HQPlayer user guide, it is mentioned "When an impulse response file is selected through “Browse...”, it's estimated gain

function is calculated and displayed in “IR gain” box". In my case, nothing is displayed in the IR Gain box...

 

Any idea?

 

Thank you!

[tboooe]

1 hour ago, satbox said:

In the HQPlayer user guide, it is mentioned "When an impulse response file is selected through “Browse...”, it's estimated gain

function is calculated and displayed in “IR gain” box". In my case, nothing is displayed in the IR Gain box...

I just checked and I do not have anything in the IR Gain box either.  However, if I select the wav files again I see the number in the IR Gain box, if I exit out of the convolution menu then go back, the IR Gain box is empty again.  I created mono L R wav files using REW.

 

@Miska any ideas?  Does this mean that convolution is not happening?

[ted_b]

My IR Gain box doesn't show currently, but convolution (in my case a seldom used 78hz notch filter; too much veil likely due to lower rez filter files) is easily recognized as being in effect.  The best way to decide on IR gain is to look at the log file and choose a gain that is the negative of what the log file says.

[tboooe]

12 minutes ago, ted_b said:

The best way to decide on IR gain is to look at the log file and choose a gain that is the negative of what the log file says.

Good suggestion Ted!   The simpleton's method is to decrease the Gain Comp until you dont see any more clipping (i.e. Limited) in HQP.

 

 Still dont understand why the IR Gain box is empty after initially selecting the wav files.

[scan80269]

On 8/22/2017 at 8:14 PM, elan120 said:

It is still inconclusive at this point.  I tried using coax1 out from SU-1 to Spring DAC coax1 in, per your suggestion, and that didn't make any difference.  I then again move back to I2S connection, and it will play 384K all the way to Spring DAC, but when HQP playing 352.8K, SU-1 does show signal came in (middle blue LED on), but no signal out to Spring DAC (LED showing ---- ---- ---- ---- ----).

 

Without knowing more about SU-1 circuit, it is getting a bit more challenging to continue.  I did measure SU-1 22.5792MHz oscillator pin-1 and pin-3 output, when I play Foobar at 44.1K and HQP at 352.8K output, pin-1 will go from low to high (0VDC to 3.3VDC), and pin-3 will have output (1.6x VDC), and same result also applied to 24.576MHz oscillator when playing the 48K output families.  This shows that incoming portion of SU-1 is working correctly, but I am a bit lost where the signal goes after that to continue measure and trace the signal, especially why 44.1K and 352.8K output signal isn't going to the I2S output when the output from oscillator (pin-3) is correct.

As of yesterday I have fixed @elan120's SU-1.  It had an issue of the 22.5792MHz reference clock being cut off from the Xilinx FPGA.  A small resistor in series with the clock trace sat crooked on the circuit board with one pad making intermittent contact which eventually became an open circuit hard failure.  After resoldering this resistor into position this SUI-1 unit has regained full functionality in playing 44.1K-based PCM content up to 352.8K as well as DSD up to DSD512.

 

[elan120]

7 minutes ago, scan80269 said:

As of yesterday I have fixed @elan120's SU-1.  It had an issue of the 22.5792MHz reference clock being cut off from the Xilinx FPGA.  A small resistor in series with the clock trace sat crooked on the circuit board with one pad making intermittent contact which eventually became an open circuit hard failure.  After resoldering this resistor into position this SUI-1 unit has regained full functionality in playing all content with 44.1K family sample rate as well as DSD up to DSD512.

 

Special thanks to @scan80269 for finding and repaired this issue.  The crooked sitting resistor came originally from the factory that has been working since I started using this SU-1 several months ago, but @scan80269 was able to spot the problem quickly and get the problem resolved.

 

I feel very blessed to have this kind of help and assistance!  My SU-1 is now playing great music using HQPlayer upsampled to DSD512 since received it back from  @scan80269  yesterday.  

[Miska]

On 8/27/2017 at 4:54 PM, k6davis said:

Unlike what others are experiencing, my system will hang at the end of a track, randomly, regardless of the format. I can put a PCM file on repeat and it will hang at the end after a few plays. The fact that the Roon/HQP combo doesn't have this problem with the older version of HQP suggests to me that the issue is on the HQP side - if it's not something in my configuration, which is of course a possibility. I'd like to stay current with HQP,  but not with the interruptions. 

 

I'm suspecting Roon is incorrectly assuming the playback position time from HQPlayer to be reliable and to ever reach end of track, which it is not and which it may not. It is plainly informative. Only the playback state is reliable and that's the only information Roon should be looking at. In fact Roon should queue next item to HQPlayer queue way before the current item ends. But it doesn't do that. But of course I cannot know what exactly Roon is doing with the data it gets, I only know how the HQPlayer control protocol works at HQPlayer side, not how other applications use it.

[evalon]

Hi again,

 

Some days ago I posted this question (pls. see below) and notice that there have been no replies.

 

FYI I am designing a high sample rate D/A converter and it would be very useful to know if I could basically optimize it for one sampling frequency (technically and sound-wise most feasible) - and have HQPlayer in combination with Roon, Muso, JRiver, or ?? manage the   streaming and resampling   from any source incl. youtube, Tidal, or some other internet source. Thus my previous question (below) ... I would still appreciate some feedback so hopefully ok to "bump" with the question again ... 

 

Cheers & thanks for any insights - Jesper 

 

Quote

Hi all,

 

I hope it is ok that I just briefly jump in on this thread with a question ... (haven't seen it answered doing a search) ...

 

Seeing that computer audio is really evolving I wonder if it is now possible to use HQPlayer to stream music from youtube, Tidal, or some other internet music source? Could e.g. be via Roon, Muso, or Jriver ...

 

Might one of you know if this is now possible - and also reasonably simple to do (I will have a suitably fast PC & DAC but hope I won't need an NAA or the like - so as to keep things simple). 

 

I would much appreciate help in this ;-)

 

Cheers,

 

Jesper 

 

[k6davis]

16 hours ago, Miska said:

 

I'm suspecting Roon is incorrectly assuming the playback position time from HQPlayer to be reliable and to ever reach end of track, which it is not and which it may not. It is plainly informative. Only the playback state is reliable and that's the only information Roon should be looking at. In fact Roon should queue next item to HQPlayer queue way before the current item ends. But it doesn't do that. But of course I cannot know what exactly Roon is doing with the data it gets, I only know how the HQPlayer control protocol works at HQPlayer side, not how other applications use it.

 

You know roughly a million times more about this than I Miska, but whatever the shortcomings of Roon's API, it does work perfectly with HQP 3.15.1 but not with HQP 3.17.0.

 

It doesn't seem that many others are having the issue, although clipper said he also had to revert to the older HQP version for the same reason. 

 

Is there any info I could provide you to help figure it out? I'd be willing to discuss it in PM's if no one else is having the issue.

[Miska]

3 hours ago, k6davis said:

You know roughly a million times more about this than I Miska, but whatever the shortcomings of Roon's API, it does work perfectly with HQP 3.15.1 but not with HQP 3.17.0.

 

I don't know about Roon's API. I only know about HQPlayer's API.

 

Quote

Is there any info I could provide you to help figure it out?

 

Having a sequence to reproduce this issue with hqp-control command line utility would certainly help...

[Miska]

12 hours ago, evalon said:

Some days ago I posted this question (pls. see below) and notice that there have been no replies.

 

FYI I am designing a high sample rate D/A converter and it would be very useful to know if I could basically optimize it for one sampling frequency (technically and sound-wise most feasible) - and have HQPlayer in combination with Roon, Muso, JRiver, or ?? manage the   streaming and resampling   from any source incl. youtube, Tidal, or some other internet source. Thus my previous question (below) ... I would still appreciate some feedback so hopefully ok to "bump" with the question again ... 

 

Yes, you can do that. Not much different from my DSC1 DAC design, which is optimized for DSD256+ but works also at DSD128.

 

For this kind of use case I would recommend using HQPlayer Embedded, since it allows you to use both analog (ADC) and S/PDIF / AES inputs with suitable hardware for wider range of sources (such as Google's Chromecast Audio).

 

[evalon]

@Miska:

 

Quote

Yes, you can do that. Not much different from my DSC1 DAC design, which is optimized for DSD256+ but works also at DSD128.

 

For this kind of use case I would recommend using HQPlayer Embedded, since it allows you to use both analog (ADC) and S/PDIF / AES inputs with suitable hardware for wider range of sources (such as Google's Chromecast Audio).

 

Thanks Miska. Will just consider what to do.

 

Have a good day 

 

Jesper

[Le Concombre Masqué]

On 29/8/2017 at 10:09 PM, Miska said:

 

Yes, you can do that. Not much different from my DSC1 DAC design, which is optimized for DSD256+ but works also at DSD128.

 

For this kind of use case I would recommend using HQPlayer Embedded, since it allows you to use both analog (ADC) and S/PDIF / AES inputs with suitable hardware for wider range of sources (such as Google's Chromecast Audio).

 

Miska, Could you please provide details about Embedded (yes I visited signalyst.com )?

 

I read it accepts analog signals ; does it mean that it would do analog to digital conversion of a vinyl flux ? (+ live room correction of LPs, as you already mentioned.)

 

What kind of "box" would then accept xlr connections from my phono pre, feed Embedded, xlr output a digital room corrected flux to a TEAC 501 ?

 

 

[steveoat]

Why does HQ Player not install into my X86 Program Folder?

 

I am running Windows  10, 64 bits.  Whenever I install HQ Player, it installs to the Program Files rather than the Program Files (X86) folder, which I assume a 64 bit program would default to.

 

What am I doing wrong?

[ted_b]

Nothing.  Frankly, I would install it to the root C drive.

[scan80269]

51 minutes ago, steveoat said:

Why does HQ Player not install into my X86 Program Folder?

 

I am running Windows  10, 64 bits.  Whenever I install HQ Player, it installs to the Program Files rather than the Program Files (X86) folder, which I assume a 64 bit program would default to.

 

What am I doing wrong?

There's nothing wrong like Ted said.

 

By default, Windows installs 64-bit apps under "Program Files" folder and 32-bit apps under "Program Files (x86)" folder.  Since HQPlayer will install 64-bit edition for Windows 10 64-bit OS, "Program Files" is the correct destination folder.

 

[Miska]

7 hours ago, Le Concombre Masqué said:

Miska, Could you please provide details about Embedded (yes I visited signalyst.com )?

What kind of details you are after?

 

7 hours ago, Le Concombre Masqué said:

I read it accepts analog signals ; does it mean that it would do analog to digital conversion of a vinyl flux ? (+ live room correction of LPs, as you already mentioned.)

 

Yes, you can use either digital (S/PDIF / AES/EBU) sources, such as CD spinner, Chromecast Audio, Apple Airport Express or similar. You can also use ADC for analog sources.

 

7 hours ago, Le Concombre Masqué said:

What kind of "box" would then accept xlr connections from my phono pre, feed Embedded, xlr output a digital room corrected flux to a TEAC 501 ?

 

I have two RME interfaces attached to my machine, RME HDSPe AIO and RME ADI-2 Pro for the purpose. ADI-2 Pro can be used at 768 kHz. It also supports DSD256 as ADC and I have all the necessary code for it, but I still need to enable that support in HQPlayer Embedded (needs some QA testing). HDSPe AIO has also ADC, but at 192/24.

 

TEAC UD-501 unfortunately is not so great with Linux... Newer UD-503/NT-503 are fine.

 

[Le Concombre Masqué]

26 minutes ago, Miska said:

What kind of details you are after?

 

 

Yes, you can use either digital (S/PDIF / AES/EBU) sources, such as CD spinner, Chromecast Audio, Apple Airport Express or similar. You can also use ADC for analog sources.

 

 

I have two RME interfaces attached to my machine, RME HDSPe AIO and RME ADI-2 Pro for the purpose. ADI-2 Pro can be used at 768 kHz. It also supports DSD256 as ADC and I have all the necessary code for it, but I still need to enable that support in HQPlayer Embedded (needs some QA testing). HDSPe AIO has also ADC, but at 192/24.

 

TEAC UD-501 unfortunately is not so great with Linux... Newer UD-503/NT-503 are fine.

 

thank you; very interesting possibilities, maybe even amazing :  Do I understand correctly that with Embedded it won't be mandatory to have a AD converter to convert the spinning on turntable LP flux, provided one has a Linux & DSD 256 compatible DAC ?

[ted_b]

DSD (whatever the bit rate) is seen by a DAC as digital.  So LP to DSD requires ADC in their somewhere.  Not sure what LP flux is, but you'll need a needledrop ripper of some sort.  But you know all this already, so maybe I'm missing your point.