HQ Player

[bogi]

I have now added "Auto" as first item to sample rate and mode drop-lists in main window. Down side with the current GUI design is that you don't see what ends up being selected inside the engine. But it is less issue for those DACs/NAAs that have front panel display.

 

Thanks, Miska! I will test it when it will be available.

[Miska]

Are there any plans to make the Invert Phase & Repeat single/all functions accessible from the API?

 

I can add those...

[Pete86]

Hi All,

 

Apologies if this has been discussed, but I could not find anything on this topic using search. I am using HQPlayer trial with my Devialet 120 (which is compatible with DSD64 over DoP). I notice that when output is set to SDM(DSD), using any of the filters/modulators, the volume is lower by a few dB as compared to PCM output. I have found this to be the case when running HQPlayer on a WS2012 PC connected to my DAC, and also when using NAA on the WS2012 PC and running HQPlayer on another Windows 10 PC on my network. Is this normal behavior? I note that the volume dial appears to be inactive when in DSD output mode.

 

This is my first experience with DSD upsampling and I have no experience with DSD files, so I apologize if this is a silly question.

 

Thanks!

[bogi]

I also recommend just taking the original DXD file (not the MQA variant) and playing that as upsampled to highest rate DSD your DAC can support for comparison.

 

Previously I found that the 96k recording sounded me lifeless and less involving compared to the MQA file. Today I could compare both with the original DXD file.

 

I did this comparison:

DXD 24/352.8: poly-sinc ASDM7 6144000 SDM(DSD)

PCM 24/96: poly-sinc ASDM7 6144000 SDM(DSD)

MQA 24/44.1: poly-sinc-mqa ASDM7 5644800 SDM(DSD)

 

What really surprised me was the difference between the original DXD master and the downsampled 96k file. I didn't expect so big difference. As HQPlayer and Foobar2000/SoX user I knew what can do upsampling. But the amount of information lost in the downsampling process from 352.8k to 96k still surprised me. Did they use really good downsampling filters? Hopefully yes.

 

The DXD recording contained much more air between instruments and sounded vivid. The subtle detail in the 96k recording was less audible. The 96k recording didn't contain so much and air and sounded me darker, smothered, lifeless in comparison with the DXD recording. The presentation of highest part of audible frequencies seemed me to be much better on the DXD master.

 

Then I compared the MQA file with the DXD file. The MQA one missed the subtle detail and clarity of the DXD master. With the DXD master I had natural feeling that the air is "right". At least I couldn't compare it with better recording, the "air" was here and sounded me naturally. The MQA recording somehow attempts to be airy, but doesn't have the same resolution as the DXD master. Therefore it's "air" sounded me somehow artificial and little disrupted in comparison with the DXD master. I perceived less low level detail, less clarity, less natural presentation of the MQA file.

 

Listening impressions are always subjective and related to the listener and his gear. Please take my post this way.

[bogi]

Pete86, 2 checkboxes in DSDIFF/DSF Settings dialog affect that:

6dB setting ... search for "-6" here http://www.computeraudiophile.com/f8-general-forum/hqplayer-resampling-filter-setup-guide-ordinary-person-13298/#post177187

DirectSDM ... search for "-3 dB" here http://www.computeraudiophile.com/f11-software/hq-player-20293/index14.html#post359633

[jhwalker]

Previously I found that the 96k recording sounded me lifeless and less involving compared to the MQA file. Today I could compare both with the original DXD file.

 

I did this comparison:

DXD 24/352.8: poly-sinc ASDM7 6144000 SDM(DSD)

PCM 24/96: poly-sinc ASDM7 6144000 SDM(DSD)

MQA 24/44.1: poly-sinc-mqa ASDM7 5644800 SDM(DSD)

 

What really surprised me was the difference between the original DXD master and the downsampled 96k file. I didn't expect so big difference. As HQPlayer and Foobar2000/SoX user I knew what can do upsampling. But the amount of information lost in the downsampling process from 352.8k to 96k still surprised me. Did they use really good downsampling filters? Hopefully yes.

 

The DXD recording contained much more air between instruments and sounded vivid. The subtle detail in the 96k recording was less audible. The 96k recording didn't contain so much and air and sounded me darker, smothered, lifeless in comparison with the DXD recording. The presentation of highest part of audible frequencies seemed me to be much better on the DXD master.

 

Then I compared the MQA file with the DXD file. The MQA one missed the subtle detail and clarity of the DXD master. With the DXD master I had natural feeling that the air is "right". At least I couldn't compare it with better recording, the "air" was here and sounded me naturally. The MQA recording somehow attempts to be airy, but doesn't have the same resolution as the DXD master. Therefore it's "air" sounded me somehow artificial and little disrupted in comparison with the DXD master. I perceived less low level detail, less clarity, less natural presentation of the MQA file.

 

Listening impressions are always subjective and related to the listener and his gear. Please take my post this way.

 

Thank you for sharing your impressions. Will be very interesting when MQA-capable DACs are available so we can do a proper comparison between DXD and MQA file "decoded", instead of just the truncated 16-bit version

[zorntel]

Jussi,

 

Not sure whether you are seeing this over on the Roon community or not:

 

KMan

2h

 

Regarding DSD playback stopping, I am seeing a very repeatable pattern .

 

• While playing DSD64 (selecting the album and using Play Album -> Play Now), the time/waveform slider stops on Roon exactly after 12 minutes and 38 seconds.
  
• But music continues to play, it finishes playing the current track and plays the next track too. Then stops.
  
• If you ask Roon to skip forward, It skips two tracks.
  

Previously, I thought the music stops where the time slider stopped, but looking closer, I realized it is always after a fixed amount of time. Would be interesting to see if this time is specific to my setup or anyone else is seeing similar behavior. 12 minutes 38 seconds of stereo DSD64 is approximately half a gigabyte of uncompressed PCM, if that rings any bells to the devs re buffer size.

I was able to repeat the above every time. Both Roon and HQPlayer on Mac- >playing to NAA -> Hugo TT. Hqplayer (3.13.0b2, DSD->PCM 384KHz or direct DSD)

 

Reply

 

 

 

Reply as linked Topic

 

 

zorntelRobert Morrison

1h

 

ooo...fun KMan. Mine just stopped once again (I'm back to running 3.12 incidentally). I'm going to run it with a long track so its easy to keep an eye on the timing. I'll also run it with a DSD128 file...that should fail twice as fast if it is a buffer issue as you suggest.

 

Reply

 

 

 

Reply as linked Topic

 

 

zorntelRobert Morrison

9m

 

As anticipated Roon froze at 6:20 on a DSD128 Opus3 DSF file...Kman QED! I had only loaded a track that was about 7 minutes...it played out in HQplayer...but roon's progress bar was frozen right at 6:20. Then I loaded a longer DSD128 track (Reference Recordings DSF file) and once again Roon froze at exactly 6:20. As KMan reported playback continued all the way through the 13:54 track and went onto the next track. Note...Roon was still showing that it was frozen on the prior track at 6:20. The next DSD128 track was 8:31 and it played to its conclusion (which was a total of 22:26 in HQplayer). Playback then stopped in spite of another track in the Roon queue. Now onto DSD256 for more fun and faster tests.

HQPlayer on Mac Mini > playing to NAA > Auralic Vega. HQplayer (3.12) Roon build 99. Output in HQ player was DSD128 (5644800)

 

 

 

 

 

[ericuco]

Could you guys looking at the Roon integration issues, please use:

 

http://www.computeraudiophile.com/f11-software/roon-hqplayer-integration-26975/ thread

[madfloyd]

Hi all, I'm wondering if someone can give me some advice.

 

I have a CAPS v3 with an (i7 3770 - I'm guessing this is 3rd generation i7).

 

I am running Roon Server and HQPlayer and upsampling all PCM -> DSD 256.

 

I was surprised to find that my server could handle the up sampling of not just redbook but 24/192 to DSD 256 using any filter. That said, the CPU load is anywhere between 38% - 47% depending on the source file.

 

Would I get any benefit from upgrading to, say, a 6th generation i7 processor? Or put another way, does keeping the CPU load really low make a difference?

 

Something that may be material is that I go out from server via ethernet and convert to fiber for the run to my listening room and then back to ethernet into a NADAC. So no USB in this scenario.

 

Thanks in advance!

[zorntel]

Could you guys looking at the Roon integration issues, please use:

 

http://www.computeraudiophile.com/f11-software/roon-hqplayer-integration-26975/ thread

 

sorry guys...just trying to make sure Jussi saw this as its on his to do list for 3.13.

[firedog]

I have now added "Auto" as first item to sample rate and mode drop-lists in main window. Down side with the current GUI design is that you don't see what ends up being selected inside the engine. But it is less issue for those DACs/NAAs that have front panel display.

 

What is the significance of the "auto" item? I don't understand...

 

BTW, with 3.13b3 DSD>PCM is now working for me flawlessly. In the previous version (b2), it wasn't working.

[bogi]

What is the significance of the "auto" item? I don't understand...

 

It's needed only for the case (all must be true):

- you are using playlist (it means you are not using album transport)

- your playlist contains tracks with mixed sample rates from both 44.1k and 48k sample rate families

- you have set an upsampling filter which can do only integer multiple of sample rate, for example closed form or poly-sinc-mqa

 

More details here and in some following posts:

http://www.computeraudiophile.com/f11-software/hq-player-20293/index183.html#post505043

[bogi]

Quote from the neighbor thread:

Did anything change on "none" PCM filter meaning? Did you mean change of the case: DirectSDM enabled, PCM defaults 'Filter'=none, PCM input and SDM(DSD) output?

http://www.computeraudiophile.com/f11-software/hq-player-20293/index82.html#post424353

 

I just tested the new beta, so I can answer myself. The special meaning of PCM defaults 'Filter'=none was removed. Filter none in PCM defaults has now no effect if output is set to SDM(DSD). Previously it had special meaning "Play PCM as PCM, DSD as DSD".

 

The most right combo box, where you can choose PCM or SDM(DSD), contains now a new item Auto. If that one is chosen, PCM file is played as PCM and DSD as DSD.

 

Another Auto can be chosen in the sample rate selection combo box. It works as it was previously discussed in this thread. In the case of non apodizing upsampling filters, which can do only integer multiple of sample rate, it automatically selects the max. possible sample rate within the sample rate family of the track. It worked for me fine, I found no issues.

 

Miska corrected also the issue when HQPlayer was restarted with non empty playlist and some tracks were added to playlist.

 

Thanks!

[Hammer]

Hi all, I'm wondering if someone can give me some advice.

 

I have a CAPS v3 with an (i7 3770 - I'm guessing this is 3rd generation i7).

 

I am running Roon Server and HQPlayer and upsampling all PCM -> DSD 256.

 

I was surprised to find that my server could handle the up sampling of not just redbook but 24/192 to DSD 256 using any filter. That said, the CPU load is anywhere between 38% - 47% depending on the source file.

 

Would I get any benefit from upgrading to, say, a 6th generation i7 processor? Or put another way, does keeping the CPU load really low make a difference?

 

Something that may be material is that I go out from server via ethernet and convert to fiber for the run to my listening room and then back to ethernet into a NADAC. So no USB in this scenario.

 

Thanks in advance!

 

 

From my experience, adding a recent CUDA card is an easier solution if processing power is the issue...

[earflappin]

Not sure if this has already been covered or not, but would love to hear from Miska Alexey as to the significance of the closed form filter touted by Schiit Audio that appears to heavily influence the sonic performance of their Yggdrasil and other multi-bit DACs.

 

This closed form filter is said to preserve the original samples, adding new interpolated samples in between, and optimizing for both frequency and time.

 

I know HPQ has closed form filter options, but IIRC, Miska himself or perhaps others I have seen post, tend not to prefer the closed form filters.

 

Thanks.

 

P.S. Is anyone using HPQ on a Mac with a MH Firewire DAC? I tried HPQ in my set up the other night without success. Will try again today.

[Chodi]

Only tried this new b3 version on closed form/auto/dsd and it resulted in dsd 256 with significantly lower cpu and gpu usage from the previous build. This is a very welcome improvement. I will try other combinations later today to see if this reduction in load is across the board. Very impressive work as always.

[Forehaven]

Not sure if this has already been covered or not, but would love to hear from Miska Alexey as to the significance of the closed form filter touted by Schiit Audio that appears to heavily influence the sonic performance of their Yggdrasil and other multi-bit DACs.

 

This closed form filter is said to preserve the original samples, adding new interpolated samples in between, and optimizing for both frequency and time.

 

I know HPQ has closed form filter options, but IIRC, Miska himself or perhaps others I have seen post, tend not to prefer the closed form filters.

 

Thanks.

 

P.S. Is anyone using HPQ on a Mac with a MH Firewire DAC? I tried HPQ in my set up the other night without success. Will try again today.

 

I still have my LIO 8 and I just listened to it the other day for pcm and using HQP without problems. Did you try turning the LIO on/off to see if it'll pick up the signal? Make sure your MIO settings are correct too. Just thinking off the top of my head

[earflappin]

I still have my LIO 8 and I just listened to it the other day for pcm and using HQP without problems. Did you try turning the LIO on/off to see if it'll pick up the signal? Make sure your MIO settings are correct too. Just thinking off the top of my head

 

I sure did thank you ! The issue was that I use VNC to the Mac Mini and my large music library was causing a lot of latency. Once I got past that I got it working fine. I tried the Poly sinc and then the closed form briefly up sampling my red book stuff to 176.4khz. To my ears - and it was a brief listen - the closed form produced a level of performance that I heard as more coherent. I am excited to experiment more

 

What filter do u prefer ? I pinged Metric Halo earlier to see if their coming 3D upgrade will provide enhanced sonics.

[Rizlaw]

I read that HQP has MPRIS support. I would like to control HQP with an on screen applet called "Now Playing" which is found in the "Screenlets" program under Linux. The "Now Playing" applet requires three fields to be filled in with information regarding the media player to be controlled via MPRIS. Below is a sample of the information needed to control the "Guayadeque" player in Ubuntu 14.04LTS.

 

Remote application: org.mpris.guayadeque

Remote object : /Player

Interface function : org.freedesktop.MediaPlayer

 

Does anyone using a Linux OS know what information to input into these three fields to get control of HQP?

 

Thanks for any help.

[zackthedog]

I have a question I probably know the answer to but wanted to check. I wanted to repurpose an old Mac Mini PowerPC to use as an NAA. I installed Debian but cannot get any of the networkaudiod packages to install. Will they not work on this machine?

[Chodi]

I just want to point out that on this latest b3 beta the auto function does not seem to work as intended with my windows based system. My dsd dac does not support fractional settings and the auto setting does not seem to discover the correct optimum setting for my equipment. Just pass this along as I realize it is beta and comments may help in the development of the final version.

[bogi]

My dsd dac does not support fractional settings and the auto setting does not seem to discover the correct optimum setting for my equipment.

 

Hi Chodi, what does it mean 'fractional settings' ? Does it mean that your DAC doesn't suport 3.1 and 6.1, but supports 2.8 and 5.6?

[Chodi]

Hi Chodi, what does it mean 'fractional settings' ? Does it mean that your DAC doesn't suport 3.1 and 6.1, but supports 2.8 and 5.6?

 

Yes that's what it means.

[bogi]

Yes that's what it means.

 

Then isn't solution for you to set limit to 5.6 (or 11.2 if your DAC can do DSD256) in the PCM defaults of the Settings dialog?

[Chodi]

Then isn't solution for you to set limit to 5.6 (or 11.2 if your DAC can do DSD256) in the PCM defaults of the Settings dialog?

 

Yes that works without issue as long as you select a filter that can do the conversion. My point is that the auto setting is of no value to those without a dac that can do all rates as it will pick rates not compatible. Not a complaint just an observation.