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9 hours ago, seatrope said:

Yes Hammer, I have it running under server 2016 core, I even have CUDA offload working in core. Just make sure to install Audiophile Optimizer first as it installs some required stuff in its WASAPI support package I think that allows hqplayer to run.

 

Hi Seatrope, is there any special procedure to follow? Thanks for your advice in advance.

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19 hours ago, lucretius said:

 

In my expereience with amateur radio, a "cheap" clock (used for frequency determination) should stabilize (be relatively drift free) within an hour and a "good" clock (e.g. oven controlled) should stabilize under a minute.  I don't know where you are getting the 24-48 hours from.

 

Don't remember where, and I'm not going to take the time to look, but a few years ago there were  measurements published showing (If memory is correct) that digital clocks in audio gear can take at least several hours to be truly stable, in some models longer than that.

Maybe tedb remembers where this info came from.  

Main listening (small home office):

Main setup: Surge protectors +>Isol-8 Mini sub Axis Power Strip/Protection>QuietPC Low Noise Server>Roon (Audiolense DRC)>Stack Audio Link II>Kii Control>Kii Three BXT (on their own electric circuit) >GIK Room Treatments.

Secondary Path: Server with Audiolense RC>RPi4 or analog>Cayin iDAC6 MKII (tube mode) (XLR)>Kii Three BXT

Bedroom: SBTouch to Cambridge Soundworks Desktop Setup.
Living Room/Kitchen: Ropieee (RPi3b+ with touchscreen) + Schiit Modi3E to a pair of Morel Hogtalare. 

All absolute statements about audio are false :)

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23 hours ago, ted_b said:

And even then, most well designed digital units that use tubes have a standby operation that keeps the clocks on and simply turns off the tube section.

As far as I can tell the Nagra does not offer a stand-by mode... :(

 

NUC10i7 + Roon ROCK > dCS Rossini APEX DAC + dCS Rossini Master Clock 

SME 20/3 + SME V + Dynavector XV-1s or ANUK IO Gold > vdH The Grail or Kondo KSL-SFz + ANK L3 Phono 

Audio Note Kondo Ongaku > Avantgarde Duo Mezzo

Signal cables: Kondo Silver, Crystal Cable phono

Power cables: Kondo, Shunyata, van den Hul

system pics

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6 hours ago, auricgoldfinger said:

I am interested in parametric equalization of my headphones.  Is there a plug-in or some other way to do parametric EQ in HQPlayer?

 

See my post two days ago.

 

(Too bad the new forum platform omits post numbers to facilitate referring to a specific prior post.  Even date and time is dicey because most of us use local time instead of GMT.)

HQPlayer (on 3.8 GHz 8-core i7 iMac 2020) > NAA (on 2012 Mac Mini i7) > RME ADI-2 v2 > Benchmark AHB-2 > Thiel 3.7

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16 hours ago, Bob Stern said:

 

See my post two days ago.

 

(Too bad the new forum platform omits post numbers to facilitate referring to a specific prior post.  Even date and time is dicey because most of us use local time instead of GMT.)

 

Please let us know how you implement the PEQ once you have done it.  Thanks!

 

 

 

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On 4/18/2017 at 11:46 AM, auricgoldfinger said:

I am interested in parametric equalization of my headphones.  Is there a plug-in or some other way to do parametric EQ in HQPlayer?

Sure! I use REW to create a set of filters, then export them as impulse response WAV files for both channels. These are then loaded directly into HQPlayer convolver. The filters are applied automatically when HQPlayer is playing, as long as you have the convolver enabled. Let me know if you need any specific step described in more detail.

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I have an occasional issue with the SDM playback.
My setup is: Roon+HQPlayer (3.16.1) -> Vinnie Rossi LIO

When I turn my PC off and turn it back on, I cannot make HQPlayer use SDM anymore. In Roon it appears the music is playing but the progress bar is not moving and of course there is no sound out. There are also some random things I do which turns SDM off, but cannot tell what exactly happens. 


If I click on Pause in Roon, change to PCM in HQPlayer the music is playing. If I again click on Pause, switch back to SDM and click play, nothing. 
The only thing that helps is a reboot off all devices, for which I am not sure of sequence yet (sometimes I need to reboot multiple times). 
When this issue happens, the Auto tab in HQPlayer doesn't show any bit rate when I click on it, but I can still choose between PCM and SDM. 

One of the log lines say:
2017/04/20 08:17:53 No suitable output rate for 44100, stop ??? Not sure why this is a case,

Could someone check the log and give me a suggestion what is actually wrong, please?
08.17 is when I tried with SDM
08.19 is when I tried with PCM

 

To sum it up, the issues is only with SDM not PCM. As if the auto rate gets blocked.
 

HQPlayer.log

Vinnie Rossi LIO (AVC/Tubestage, AMP Module with built in HPF 100Hz 24dB/octave, DAC 2.0), Harbeth P3ESR, Rythmik F8

Win10 i7-7700 -> Roon -> HQPlayer DSD512- > LIO 100Hz HPF -> Harbeth P3ESR

                                                                                ->LIO  -> miniDSP <100Hz -> Rythmik F8  

 

 

 

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3 hours ago, mirekti said:

One of the log lines say:
2017/04/20 08:17:53 No suitable output rate for 44100, stop ??? Not sure why this is a case,

To sum it up, the issues is only with SDM not PCM. As if the auto rate gets blocked.

 

Since the interface can do only 44.1k and 48k and you are trying to output DSD, it is not going to work...

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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10 minutes ago, Miska said:

 

Since the interface can do only 44.1k and 48k and you are trying to output DSD, it is not going to work...

 

I am not sure I understand what is going on though. It works fine until something in the chain gets broken.
Does this mean the communication between the DAC and HQPlayer gets broken or what?
I mean, I can still choose SDM or PCM in HQPlayer. 
BTW this configuration works fine when I give it a "fresh" start.

Vinnie Rossi LIO (AVC/Tubestage, AMP Module with built in HPF 100Hz 24dB/octave, DAC 2.0), Harbeth P3ESR, Rythmik F8

Win10 i7-7700 -> Roon -> HQPlayer DSD512- > LIO 100Hz HPF -> Harbeth P3ESR

                                                                                ->LIO  -> miniDSP <100Hz -> Rythmik F8  

 

 

 

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1 hour ago, mirekti said:

 

I am not sure I understand what is going on though. It works fine until something in the chain gets broken.
Does this mean the communication between the DAC and HQPlayer gets broken or what?
I mean, I can still choose SDM or PCM in HQPlayer. 
BTW this configuration works fine when I give it a "fresh" start.

 

I'm not sure I understand either. :)

 

At startup:

2017/04/20 08:17:38 SDM packing: 1

 

You have DoP enabled.

 

  2017/04/20 08:17:38 Loaded ASIO driver: XMOS USB Audio 2.0 ST 2023
  2017/04/20 08:17:38 ASIO default format is PCM
  2017/04/20 08:17:38 Rate available: 44100
  2017/04/20 08:17:38 Rate available: 48000
  2017/04/20 08:17:38 Default ASIO channels: 0 in / 2 out

 

So the driver supports only these two PCM rates... Then you switch to SDM mode:

 

  2017/04/20 08:17:38 Audio engine SDM mode enabled

 

But the problem is that there are no legal DSD output rates. With DoP; 44100*16 = 705600 which HQPlayer doesn't allow as DSD sample rate. Neither 48000*16 = 768000.

 

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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1 hour ago, Miska said:

 

But the problem is that there are no legal DSD output rates. With DoP; 44100*16 = 705600 which HQPlayer doesn't allow as DSD sample rate. Neither 48000*16 = 768000.

 

 

I am not sure I am able to follow you. Do I have something wrong in my settings?
Just came from the office and did the following. Power off LIO -> Reboot PC -> Power on LIO -> Launched HQPlayer -> Launched Roon -> SDM Playback OK 
Also, I can see on the LIO display if it is PCM (it would show 44) or SDM (it shows 128),
I do see some warnings in the log, though. 
The log for the above starts at line 206, time 17:15:15

 

Do you think there's an issue with the LIO drivers or the PC?

HQPlayer - Copy.log

Vinnie Rossi LIO (AVC/Tubestage, AMP Module with built in HPF 100Hz 24dB/octave, DAC 2.0), Harbeth P3ESR, Rythmik F8

Win10 i7-7700 -> Roon -> HQPlayer DSD512- > LIO 100Hz HPF -> Harbeth P3ESR

                                                                                ->LIO  -> miniDSP <100Hz -> Rythmik F8  

 

 

 

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...just a quick update. I figured out how to "reset" it. 

 

1. I need to turn LIO off.

2. Restart HQPlayer which will than complain there's no device.
3. Turn LIO back on
4. Start HQPlayer

 

Aftert I complete those four steps the playback is fine and I can see following in the log file:

ASIO default format is PCM
Rate available: 44100
Rate available: 48000
Rate available: 88200
Rate available: 96000
Rate available: 176400
Rate available: 192000
Rate available: 352800
Rate available: 384000
 

I do however wonder how all this is working as the numbers you mentioned were 705600 and 768000. 

Vinnie Rossi LIO (AVC/Tubestage, AMP Module with built in HPF 100Hz 24dB/octave, DAC 2.0), Harbeth P3ESR, Rythmik F8

Win10 i7-7700 -> Roon -> HQPlayer DSD512- > LIO 100Hz HPF -> Harbeth P3ESR

                                                                                ->LIO  -> miniDSP <100Hz -> Rythmik F8  

 

 

 

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6 hours ago, mirekti said:

...just a quick update. I figured out how to "reset" it. 

 

1. I need to turn LIO off.

2. Restart HQPlayer which will than complain there's no device.
3. Turn LIO back on
4. Start HQPlayer

 

Aftert I complete those four steps the playback is fine and I can see following in the log file:

ASIO default format is PCM
Rate available: 44100
Rate available: 48000
Rate available: 88200
Rate available: 96000
Rate available: 176400
Rate available: 192000
Rate available: 352800
Rate available: 384000
 

I do however wonder how all this is working as the numbers you mentioned were 705600 and 768000. 

Is this the LIO DAC you have? The spec says it does DSD over DoP. There is a setting for that in HQPlayer. The logs you have show your driver supports only PCM - not sure if this is correct for a DSD over DoP DAC, mine does native DSD.

 

BTW what happened to profiles with the forum upgrade?

🎸🎶🏔️🐺

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6 hours ago, mirekti said:

Aftert I complete those four steps the playback is fine and I can see following in the log file:

ASIO default format is PCM
Rate available: 44100
Rate available: 48000
Rate available: 88200
Rate available: 96000
Rate available: 176400
Rate available: 192000
Rate available: 352800
Rate available: 384000
 

I do however wonder how all this is working as the numbers you mentioned were 705600 and 768000. 

 

With those rates listed it should work up to DSD128 (possibly including 48x128)... Maybe some driver or firmware bug that causes it to lose those higher rates at some point. At least it is good to check that the device is not set as default audio output device for Windows. Multiple applications accessing the device simultaneously may cause strange behavior if the drivers are not specifically prepared to deal with such situations gracefully.

 

If you can find some systematic sequence of actions which causes the problem of lost rates, you could report that to the manufacturer.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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On 4/19/2017 at 4:29 PM, pkane2001 said:

Sure! I use REW to create a set of filters, then export them as impulse response WAV files for both channels. These are then loaded directly into HQPlayer convolver. The filters are applied automatically when HQPlayer is playing, as long as you have the convolver enabled. Let me know if you need any specific step described in more detail.

 

I've never used REW before, but I figured out the following procedure.  I'd be grateful for your advice on simplifications or improvements: 

 

Prefs > View > clear “Enable mouse wheel zoom” if using a trackpad.

 

Open EQ window by clicking EQ button.

• Gear button:  Clear “Invert Filter Responses”.  Click Gear button again to close dialog.

 

Open "EQ Filters" window by clicking EQ Filters button (top center of EQ window).

• Clear “Always on top”.

• Control > Manual.

• Type > PK for parametric EQ.

• enter Q=1.4 for BW=1 octave.  (0.67 for BW=2.)  http://www.rane.com/note170.html

• close EQ Filters window.

 

Close EQ window, or else bring the main REW window to the front via cmd-tilde.

 

File menu > Export > "Export filters impulse response as WAV".

• Clear “Normalize”.  Mono.  32-bit.  Sample rate same as imported impulse response.

• No minus sign or other non-alphanumeric characters in filename.

 

Optional: Verify frequency response of exported IR WAV file:

• Open the WAV file:  File menu > Import impulse response.

• Click the “SPL & Phase” button above the graph.

• Hover mouse near lower right of graph to reveal 20–20,000 X-axis button and click it.

• Upper left/right zoom icons separately zoom Y-axis of amplitude & phase curves.

HQPlayer (on 3.8 GHz 8-core i7 iMac 2020) > NAA (on 2012 Mac Mini i7) > RME ADI-2 v2 > Benchmark AHB-2 > Thiel 3.7

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@Miska:  Convolution question:

 

The user guide says: "When source material sampling rate differs from the impulse sampling rate, impulse responses will be scaled to the source material's sampling rate. This can have a huge impact on CPU load ..."

 

It seems to me that HQP would resample the impulse response before it begins playing the audio, save it in memory or disk, and then begin playback by convolving the saved (resampled) impulse response with the music source.  This would cause a delay before playback begins, but it would not affect the CPU load during playback.  Am I missing something?

HQPlayer (on 3.8 GHz 8-core i7 iMac 2020) > NAA (on 2012 Mac Mini i7) > RME ADI-2 v2 > Benchmark AHB-2 > Thiel 3.7

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58 minutes ago, Bob Stern said:

 

I've never used REW before, but I figured out the following procedure.  I'd be grateful for your advice on simplifications or improvements: 

 

Prefs > View > clear “Enable mouse wheel zoom” if using a trackpad.

 

Open EQ window by clicking EQ button.

• Gear button:  Clear “Invert Filter Responses”.  Click Gear button again to close dialog.

 

Open "EQ Filters" window by clicking EQ Filters button (top center of EQ window).

• Clear “Always on top”.

• Control > Manual.

• Type > PK for parametric EQ.

• enter Q=1.4 for BW=1 octave.  (0.67 for BW=2.)  http://www.rane.com/note170.html

• close EQ Filters window.

 

Close EQ window, or else bring the main REW window to the front via cmd-tilde.

 

File menu > Export > "Export filters impulse response as WAV".

• Clear “Normalize”.  Mono.  32-bit.  Sample rate same as imported impulse response.

• No minus sign or other non-alphanumeric characters in filename.

 

Optional: Verify frequency response of exported IR WAV file:

• Open the WAV file:  File menu > Import impulse response.

• Click the “SPL & Phase” button above the graph.

• Hover mouse near lower right of graph to reveal 20–20,000 X-axis button and click it.

• Upper left/right zoom icons separately zoom Y-axis of amplitude & phase curves.

 

Hi Bob,

 

Looks correct to me. I do a few things differently, some are due to the fact that I'm trying to correct for measured room response, so I have a measured curve and a house curve I'm trying to achieve. Still, I mostly do manual filter selection to match the curves. For headphones, you'll have to experiment with the filter settings, as I'm not sure how you'd measure headphone output.

 

In the EQ window, I select 'Generic' as the equalizer type. This gives the most flexibility with shelf filters, LP, HP, etc., you are not limited to just PK.

 

When entering manual filter settings, don't forget to set the gain amount, in dB. I never set more than 6dB correction in either direction. And, as I mentioned, I use more than just PK filter type. High pass and low pass filters are extremely useful for adjusting a large portion of the curve before applying the more precise/limited PK filters. 

 

 

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47 minutes ago, Bob Stern said:

@Miska:  Convolution question:

 

The user guide says: "When source material sampling rate differs from the impulse sampling rate, impulse responses will be scaled to the source material's sampling rate. This can have a huge impact on CPU load ..."

 

It seems to me that HQP would resample the impulse response before it begins playing the audio, save it in memory or disk, and then begin playback by convolving the saved (resampled) impulse response with the music source.  This would cause a delay before playback begins, but it would not affect the CPU load during playback.  Am I missing something?

 

As far as I can tell, HQP resamples and convolves in near real-time. The only delay might be due to a small buffer, such as the one you can enable with some async audio drivers and with NAA. Other audio players such as Audiorvana+ have an option of decoding large chunks of files into memory, and then playing them from there. This does result in a potentially large delay before the start, but perhaps, a smoother playback. I don't see such an option in HQP.

 

I assume that convolution is an expensive operation from the CPU perspective. And if you are upsampling, then HQP must apply it to many more samples per second, resulting in a greater CPU load. But this is just my guess.

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11 hours ago, Bob Stern said:

It seems to me that HQP would resample the impulse response before it begins playing the audio, save it in memory or disk, and then begin playback by convolving the saved (resampled) impulse response with the music source.  This would cause a delay before playback begins, but it would not affect the CPU load during playback.

 

It is not saved anywhere as that operation is really fast...

 

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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10 hours ago, pkane2001 said:

 

Hi Bob,

 

Looks correct to me. I do a few things differently, some are due to the fact that I'm trying to correct for measured room response, so I have a measured curve and a house curve I'm trying to achieve. Still, I mostly do manual filter selection to match the curves. For headphones, you'll have to experiment with the filter settings, as I'm not sure how you'd measure headphone output.

 

In the EQ window, I select 'Generic' as the equalizer type. This gives the most flexibility with shelf filters, LP, HP, etc., you are not limited to just PK.

 

When entering manual filter settings, don't forget to set the gain amount, in dB. I never set more than 6dB correction in either direction. And, as I mentioned, I use more than just PK filter type. High pass and low pass filters are extremely useful for adjusting a large portion of the curve before applying the more precise/limited PK filters. 

 

 

 

Thanks for your posts.  This is great information!  It will take me some time to figure out what I'm doing since the process is beyond my current understanding of both HQP and now REW.

 

 

 

 

 

 

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10 hours ago, pkane2001 said:

 

Hi Bob,

 

Looks correct to me. I do a few things differently, some are due to the fact that I'm trying to correct for measured room response, so I have a measured curve and a house curve I'm trying to achieve. Still, I mostly do manual filter selection to match the curves. For headphones, you'll have to experiment with the filter settings, as I'm not sure how you'd measure headphone output.

 

In the EQ window, I select 'Generic' as the equalizer type. This gives the most flexibility with shelf filters, LP, HP, etc., you are not limited to just PK.

 

When entering manual filter settings, don't forget to set the gain amount, in dB. I never set more than 6dB correction in either direction. And, as I mentioned, I use more than just PK filter type. High pass and low pass filters are extremely useful for adjusting a large portion of the curve before applying the more precise/limited PK filters. 

 

 

 

I do have one quick question before I dive into the rabbit hole:  will my DSD files be played in native format or converted to WAV files?

 

 

 

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