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Now that hqp-control specs have been out for some time, has anyone tried to get a remote like JRemote (obviously involving JRIver somehow) or Linn Kazoo (OpenHome control point) to talk to HQPlayer desktop (I am not a candidate for the Linux expertise involved in HQPlayer embedded). Yes, I have Roon enabled but miss the simplicity of my Kazoo or JRemote.

 

I would love to control my HQPlayer with Kazoo or JRemote, is that even a possibility?

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I would love to control my HQPlayer with Kazoo or JRemote, is that even a possibility?

 

Ashok, that is precisely my question. It would be VERY cool. I am hoping that hqp-control gurus like Peter (bogi) and Geoffrey Armstrong would know a way.

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Ashok, that is precisely my question. It would be VERY cool. I am hoping that hqp-control gurus like Peter (bogi) and Geoffrey Armstrong would know a way.

 

A friend of mine - "realmessy" who sometimes is active here too - has developed a little app for Android.

A few screen captures:

 

Screenshot_2015-10-17-20-58-50.png

Screenshot_2015-11-01-18-52-05.png

Screenshot_2015-11-01-18-52-42.png

Screenshot_2015-11-01-18-52-56.png

 

"I can browse music by Composer -> Genre (Orchestral, Vocal, Chamber etc etc) -> Opera -> Artist / Year

or change the mode 'display at will (Genre, Artist, etc etc). The application connects to a server-side software developed in Python, so it can be installed anywhere - I have it on the same PC (Windows 10) with HQPlayer, but it could on a NAS where Music resides. It performs music indexing and it is responsible for connection with HQPlayer. Indexing is very fast, just over 40 seconds to perform a scan of 480 GB of flac, saving all the metadata in a NoSQL database. Once the data is saved on the db, it can do all possible and imaginable searches.

It uses TCP protocol. No playlist. It's possible to see the current playlist on the first tab. No track fwd/bwd as I do not use them."

 

He intends to add support for covers and stop there. Then he will release code on Github so someone can eventually improve on it.

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Just to be more clear, the restriction isn't so much the Mac or OS X--except for the fact that Coreaudio requires DoP to send DSD (unless you have a custom ASIO driven DAC like exaSound). It is the DoP 50% overhead required for framing. So a DoP capable USB input board/DAC (such as your Lampi when updated) that can accept the rate for 512 can be driven by a Mac/DoP setup to DSD256.

If your DAC can only go non-DoP to 256, then DoP will only take you to 128.

 

Exactly. I found this out after further scratching and an email from Lucas....

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I plan on using a Linux based NAA or streamer fronted by a Mac server using Roon/HQplayer. I have now been told that this paradigm can spit out DSD1024 with no issues....case closed. I confess my ignorance.....the only remaining issue is to find a Mac with enough horsepower that does not cost 8k like the latest Mac Pros! I need to maximize Geekbench score for the least amount of $$$

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This may be a sacrilegious question to ask in the context of this thread, but has anyone fiddled with a Phasure-type PCM (24/768) upsampling paradigm rather than DSD?

 

No it is not sacrilegious.. You can do that too, although there are not so many PCM DACs out there. (note the difference between PCM DAC and a DAC that merely accepts PCM input)

 

Part of the reason I ask is I have always loved NOS dacs and wondered how they would fair against "chipless" DSD when fed an upsampled stream....

 

You can find one set of results here, DSC1 vs Metrum Musette:

Signalyst DSC1 - Page 21 - diyAudio

 

I will post more complete results for Musette on my blog here at CA once I find time. I have all the measurements done already.

 

Both are discrete DAC implementations. Although DSC1 is not a finished product but a DIY design and work-in-progress project that keeps improving as time goes.

 

To me, the DSC1 sounds much better than Musette, but I'm surely biased. :D

 

I obtained the Musette because it can accept 352.8/384k input and is decently priced. I expect some improvements in performance with the more expensive models.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Question to all you HQ geeks: what is the minimum Geekbench score to safely allow PCM>DSD 256 or even 512 without hiccups? I need to score a new Mac to run HQ....

 

I just run my 1st DSD256 test with:

 

Processing = MacBookPro 11,4 i7 2.2GHz, El Capitan 10.11.2, HQPlayer 3.12 (no tuning)

NAA = Lenovo T410 i5, Linux Mint 17.3 (no tuning)

DAC = JLsounds AKM4490 with JLsounds I2overUSB I/F

 

PCM16/44 to DSD256 non-DoP > ~ 33-35% CPU (no pipeline SDM / closed-form / DSD257+fs)

DSD64 to DSD256 non-DoP > ~18-20% CPU (no pipeline SDM / closed-form / DSD257+fs)

 

Some comments:

 

- pipeline SDM doesn't work with my system and HQP 3.12 (unable to play any file ... dunno why)

- I experienced some rare dropouts (tried to tune buffer to 100mSec but sometimes still occur ... need to investigate)

- MacBookPro at stable 30% usage doesn't spin fans too fast, so noise is definitely good

 

Hope useful. Have a nice day. Massimiliano

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No it is not sacrilegious.. You can do that too, although there are not so many PCM DACs out there. (note the difference between PCM DAC and a DAC that merely accepts PCM input)

 

 

 

You can find one set of results here, DSC1 vs Metrum Musette:

Signalyst DSC1 - Page 21 - diyAudio

 

I will post more complete results for Musette on my blog here at CA once I find time. I have all the measurements done already.

 

Both are discrete DAC implementations. Although DSC1 is not a finished product but a DIY design and work-in-progress project that keeps improving as time goes.

 

To me, the DSC1 sounds much better than Musette, but I'm surely biased. :D

 

I obtained the Musette because it can accept 352.8/384k input and is decently priced. I expect some improvements in performance with the more expensive models.

 

Very interesting. I look forward to reading about it....

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I just run my 1st DSD256 test with:

 

Processing = MacBookPro 11,4 i7 2.2GHz, El Capitan 10.11.2, HQPlayer 3.12 (no tuning)

NAA = Lenovo T410 i5, Linux Mint 17.3 (no tuning)

DAC = JLsounds AKM4490 with JLsounds I2overUSB I/F

 

PCM16/44 to DSD256 non-DoP > ~ 33-35% CPU (no pipeline SDM / closed-form / DSD257+fs)

DSD64 to DSD256 non-DoP > ~18-20% CPU (no pipeline SDM / closed-form / DSD257+fs)

 

Some comments:

 

- pipeline SDM doesn't work with my system and HQP 3.12 (unable to play any file ... dunno why)

- I experienced some rare dropouts (tried to tune buffer to 100mSec but sometimes still occur ... need to investigate)

- MacBookPro at stable 30% usage doesn't spin fans too fast, so noise is definitely good

 

Hope useful. Have a nice day. Massimiliano

 

Thanks Massimiliano! That is very useful feedback....

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I have now made a first beta build of the upcoming 3.8.0 version. It doesn't yet contain all the things, but it has some of the most notable ones like

  • Extended hqp-control functionality
  • New matrix processing for channel routing and mixing
  • Adjustment of some of the internal thread priorities based on feedback
  • Remember more of the UI state across restarts

 

Plus of course usual bunch of various other changes (I don't even remember all, I'll have to compile a changelog for the final release announcement).

 

As before, it is recommended not to touch externally the process priority, because it will screw up the internal relative priorities of threads.

 

You can get the beta here:

http://www3.signalyst.com/bins/HQPlayerDesktop3-setup-380b1.exe

 

Source code for the hqp-control is now also available here:

https://www.signalyst.eu/bins/hqp-control-380b1-src.zip

It's been written in such way that extracting the XML protocol spec from the implementation is not too hard if someone wants to implement it in a different way.

 

I’m after some advice about the channel mixing and convolution function that I’ve not been able to find from searching this thread. My target set-up involves stereo speakers running full-range and three subwoofers each with their own low-pass filter (Channels 3-5) that are designed to smooth room response. All this needs to be time aligned.

 

To do this I’ve set-up the solution as a 5.1 system with Channels 1-2 used for the main speakers, Channels 3-5 used for the subwoofers and Channel 6 unused. Channels 1-2 are passed through a 1:1 convolution (from Acourate’s SimpleFilter). Meanwhile, Channel 3, for example, is created via using Channel 1 as Source and Channel 2 as Mix within the Pipeline tool and then adding a low-pass filter.

 

It seems as if this is, though, wrong as I’m finding that the subwoofers are quite out-of-sync with the main speakers. This is despite the main speakers running the unity filter and all speakers being time / distance aligned using the Speaker Set-Up utility. Any and all advice would be great. Thanks in advance.

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It seems as if this is, though, wrong as I’m finding that the subwoofers are quite out-of-sync with the main speakers. This is despite the main speakers running the unity filter and all speakers being time / distance aligned using the Speaker Set-Up utility. Any and all advice would be great. Thanks in advance.

 

Is the delay really equal in the unity filter compared to the low-pass filters. You could check this with something like Audacity by importing each filter as tracks to the same project and checking that the unit peaks are time aligned.

 

But it could also be that the unity filter gets truncated to shorter one in HQPlayer (-> less delay). I have to check this out and if there's such issue I'll try to fix it for upcoming 3.13 release.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Is the delay really equal in the unity filter compared to the low-pass filters. You could check this with something like Audacity by importing each filter as tracks to the same project and checking that the unit peaks are time aligned.

 

But it could also be that the unity filter gets truncated to shorter one in HQPlayer (-> less delay). I have to check this out and if there's such issue I'll try to fix it for upcoming 3.13 release.

 

Thanks, Miska,. Is it correct to assume that all channels need to be to convolved or will the account for the time-delay between channels that aren't filtered and those which are filtered? Also, I don't have access to Audacity but I will generate a 2-way filter in Accourate and use this high-pass for the mains / low-pass for the subwoofers to see if the problem remains.

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Thanks, Miska,. Is it correct to assume that all channels need to be to convolved or will the account for the time-delay between channels that aren't filtered and those which are filtered? Also, I don't have access to Audacity but I will generate a 2-way filter in Accourate and use this high-pass for the mains / low-pass for the subwoofers to see if the problem remains.

 

I can confirm that if I use the low- and high-pass from a 2-way filter - as opposed to the SimpleFilter and low-pass - then the timing issue goes away. Also, I've compared the SimpleFilter and Low-Pass filters within Acourate and this seems to confirm that these are time aligned so perhaps the issue is in the truncation of the SimpleFilter

 

P.S. One other observation is that I'm seeing the right channel being 3-6dB lower than the left channel despite the fact that these are both using the same, mono, filter.

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I can confirm that if I use the low- and high-pass from a 2-way filter - as opposed to the SimpleFilter and low-pass - then the timing issue goes away. Also, I've compared the SimpleFilter and Low-Pass filters within Acourate and this seems to confirm that these are time aligned so perhaps the issue is in the truncation of the SimpleFilter

 

OK, I'll check this out and fix it...

 

P.S. One other observation is that I'm seeing the right channel being 3-6dB lower than the left channel despite the fact that these are both using the same, mono, filter.

 

That shouldn't happen, there are two per-channel gain adjustments, the matrix mixing gain and then speaker setup gain.

 

Can you email me screenshots of your matrix and speaker config dialogs so I could try to reproduce the problem here?

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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OK, I'll check this out and fix it...

 

 

 

That shouldn't happen, there are two per-channel gain adjustments, the matrix mixing gain and then speaker setup gain.

 

Can you email me screenshots of your matrix and speaker config dialogs so I could try to reproduce the problem here?

 

Thanks. Email sent to [email protected]

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Jussi,

 

Just fired up 3.13.0beta2 to do some Roon build99 testing. What is poly-sinc-mqa?

 

Robert

Software: Roon/HQplayer; System I: Roon Server/HQplayer DSD 512 Upsampling, Custom Windows 10 PC/AO, LPS-1 powered Startech USB card; LPS-1 powered ISO Regen; Holo Cyan DAC; VPI Scout 2 Turntable, Soundsmith Boheme, TTW Clamps and Carbon Matt; Cary SLP-98P Preamp; Van Alstine FET 600 Poweramp;  Aerial Acoustics 6T loudspeakers, SVS SB13 Ultra Subwoofers. System II: Custom PC with Signalyst Linux HQplayer NAA; LPS-1 powered Startech USB card; LPS-1 powered ISO Regen; IFI Micro iDSD Black Label; Primaluna Dialogue 2 with Tung Sol KT-150; Paradigm Studio 20 v3 monitors on Custom Mapleshade stands. Cables: Moon Audio, LUSH, Kimber Kable, Mapleshade, LARRY custom.

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Miska,

I just got the 3.13.0beta2 and also upgraded to 99 build of Roon. When I try to play DSD64 from Roon and set HQplayer to do PCM, there is no sound and the time moves almost at double the rate. DSD->PCM works fine when used with HQplayer directly.

 

My setup: Roon (DSD64)->Hqplayer(Poly-Sinc, NS5, PCM 384K, Cuda and pipeline SDM unchecked, DirectSDM unchecked)->NAA (Mac Book Air)->Hugo TT

 

Same results for HQplayer 3.13.0beta2 on both Mac (Yosemite) and Windows 10. DSD64 or DSD64->DSD128 works fine with Roon.

 

Is there a setting in Roon or Hqplayer I am missing?

 

Thanks

Desktop: AMD Server(AL Roon Core/ HQP NAA)  >>  Fiber >> DAVE >> (2 x HFC Trinity Helix) >> LCD-5/(KGSSHV Carbon CC + CRBN)/ Omega CAMs [Everything plugged into Sound Applications TT7]

 

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Just fired up 3.13.0beta2 to do some Roon build99 testing. What is poly-sinc-mqa?

 

It is a new filter designed for playback of MQA encoded files. It of course doesn't do any decoding, but it just attempts to clean up the noise caused by MQA encoding. It is also slow roll-off and thus has very short ringing. At the moment it is a linear phase filter, but it is subject to further tuning so the properties may change.

 

Some other changes:

- Speaker delay adjustment now works also in DirectSDM mode

- Playback of DSD sources from Roon should now work properly in all cases

- Fixes some cases where Roon attempts to start playback but fails

- Now has the new shortcut keys according to the earlier discussion (function keys as marked on Apple keyboards)

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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