HQ Player

[bogi]

Question about bit perfect playback setting. Is there one?

 

Bitperfect playback is achievable with HQPlayer, but it is not the intended way how to use HQPlayer.

 

The strength of HQPlayer are software digital filters it contains. HQPlayer is intended to substitute hardware digital processing in DAC with higher quality software digital processing in HQPlayer. It is achieved by software upsampling and optional software conversion of PCM data to DSD. This is the way how HQPlayer can improve things. It simply does not sound reasonable to pay for digital filters contained in HQPlayer and then not to use them.

 

I would use bitperfect option in HQPlayer only for comparison or measurement purposes.

[Miska]

Its Sabre 9018 based dac with an Amanero usb . I am not trying to evade the question but I am certain you have not experienced my dac. This did not happen with earlier versions of your software. Here is the dac:

 

ARTT Labs | DSD Audio Professional

 

OK, I think this is a problem related to certain Amanero units. I have at least one that has similar problem at speeds of DSD256 or higher. It doesn't appear at DSD128 or DSD64. And it doesn't happen for first 10-15 seconds or so. And it happens only with right channel. I believe some people have experienced similar with Yulong DA8 too.

 

It is either a driver problem, firmware bug or component tolerance problem. But it is a known issue, I know others that have seen this too.

[Chodi]

OK, I think this is a problem related to certain Amanero units. I have at least one that has similar problem at speeds of DSD256 or higher. It doesn't appear at DSD128 or DSD64. And it doesn't happen for first 10-15 seconds or so. And it happens only with right channel. I believe some people have experienced similar with Yulong DA8 too.

 

It is either a driver problem, firmware bug or component tolerance problem. But it is a known issue, I know others that have seen this too.

 

 

Thank you for your comments. I will look into that.

[tranz]

Bitperfect playback is achievable with HQPlayer, but it is not the intended way how to use HQPlayer.

 

The strength of HQPlayer are software digital filters it contains. HQPlayer is intended to substitute hardware digital processing in DAC with higher quality software digital processing in HQPlayer. It is achieved by software upsampling and optional software conversion of PCM data to DSD. This is the way how HQPlayer can improve things. It simply does not sound reasonable to pay for digital filters contained in HQPlayer and then not to use them.

 

I would use bitperfect option in HQPlayer only for comparison or measurement purposes.

 

Hi bogi,

 

That is exactly why I am asking. What is the setting for bit perfect playback on the HQP, if there is one?

 

Cheers

[Miska]

That is exactly why I am asking. What is the setting for bit perfect playback on the HQP, if there is one?

 

When you play DSD files, it is enough to have "Direct SDM" checked in DSDIFF/DSF Settings dialog and select SDM output format.

 

For PCM, select filter as "none", dither as "none", turn volume to 0 dBFS and select PCM output format.

[bogi]

In an older posting it was mentioned setting filter and dither to none, volume to 0dB, and bit rate to default. This, however, does not provide bit perfect playback.

 

Why do you mean this does not provide bit perfect PCM playback?

[wwaldmanfan]

Hi bogi,

That is exactly why I am asking. What is the setting for bit perfect playback on the HQP, if there is one?

Cheers

 

I assume leaving the filters at their default settings (poly-sinc/TPDF), and playing a file at its native sample rate gets you as close to "bit perfect" as possible, i.e., 44100 for redbook rips, but I'm not sure. if you are not increasing the sample rate, or transcoding to DSD, do the default filters do anything?

 

The difference is small, but I think HQPlayer setup as above has slightly better SQ on my system than Audirvana Plus 1.5.10. I don't use the upsampling filters, because my DAC upconverts all incoming PCM, regardless of sample rate, to 844kHz PWM.

[Miska]

I assume leaving the filters at their default settings (poly-sinc/TPDF), and playing a file at its native sample rate gets you as close to "bit perfect" as possible, i.e., 44100 for redbook rips, but I'm not sure. if you are not increasing the sample rate, or transcoding to DSD, do the default filters do anything?

 

Those filters that allow 1:1 conversion (poly-sinc -family) are in effect also when input and output rates are the same. Practically this means performing just the apodization part of the filter without rate conversion. For poly-sinc-hb (non-apodizing) nothing much happens, somewhat depending on input data though.

 

The difference is small, but I think HQPlayer setup as above has slightly better SQ on my system than Audirvana Plus 1.5.10. I don't use the upsampling filters, because my DAC upconverts all incoming PCM, regardless of sample rate, to 844kHz PWM.

 

Usually it would be still very advantageous to convert to highest input rate supported by the DAC. The first rate increase step is the most critical one.

[tranz]

When you play DSD files, it is enough to have "Direct SDM" checked in DSDIFF/DSF Settings dialog and select SDM output format.

 

For PCM, select filter as "none", dither as "none", turn volume to 0 dBFS and select PCM output format.

 

Thanks Miska,

 

That does not provide bit perfect playback for PCM. Tried with bit rate set to default, 16, and 24 and those specific test files.

 

Not a biggie, but I was trying to establish a baseline for swapping the various filters without having to reboot and switch back to my current baseline of Amarra every time. Evaluating it for use with a discrete ladder DAC at this point. What are your fav settings if you have tried it on such a DAC? Apologies if you had already mentioned it.

 

Cheers

[Chodi]

Thanks to Miska I found the problem with very low level click. I checked the Yulong DA 8 thread and got the suggestion. The cause was my wireless router. I moved it further from the dac and the click is gone. In the process I also looked on the manufacturers web site and discovered that there is an updated driver which I installed. I guess this newer driver was to cope with Windows 8.1 and possibly Windows 10 (which I am using). At any rate, thanks to Miska I found the problem small though it was. HQP is perfect no problems.

[Jud]

Thanks Miska,

 

That does not provide bit perfect playback for PCM. Tried with bit rate set to default, 16, and 24 and those specific test files.

 

Not a biggie, but I was trying to establish a baseline for swapping the various filters without having to reboot and switch back to my current baseline of Amarra every time. Evaluating it for use with a discrete ladder DAC at this point. What are your fav settings if you have tried it on such a DAC? Apologies if you had already mentioned it.

 

Cheers

 

Not Miska, but my guess would be the highest input rate supported by the DAC, poly-sinc filters, TPDF dither?

[tranz]

Ah, good to see you over here copy_of_a! Now that you have downloaded the trial and found the preferences pane of HQP, here are a couple of tips as a starting point. Knowing you Project Box RS DAC seems to only go to 192Khz on its USB input (though it goes to DSD128 which would seem to imply that it can handle 352.8/384), try this to start:

 

For PCM, try Poly-sinc-short filter and NS4 or NS5 dither, and set sample rate limit to either 176.4 or 192. For SDM (which requires that you choose DoP as SDM Pack in pref window), try just the straight Poly-sinc and for modulator, either DSD5 or ASDM5 to start--and your max rate for SDM is going to be 5644800.

Maybe set the Volume Max to -1, -2, or -3dB--especially if you are doing conversion of Redbook to DSD.

 

After you get used to the SQ and play with the filters, maybe try setting the DAC bits to 24 (as opposed to letting HQP send 32 bits, some of the least which may get "lost" between your DACs USB interface and the rest of the DAC; I am oversimplifying here--just try it and you will know if you like it).

 

The settings you set in the HQP preferences window just become the defaults each time you launch the app. On the main screen you also have drop downs to change the filters, dither, sample rate and modulator. Doing so there just overrides the defaults for that session or track. That is handy since every time you change someone in the Preferences window it clears your playlist. So if you are experimenting with filters and SDM, do it from the main window drop downs.

 

A quick way to get playlists of tracks into HQP without using the "library" function of the program (the upper pane), is to go into iTunes, create a playlist, then highlight that playlists name (make it active that is). Then at the bottom-left, in the Playlist sidebar, click the cog icon and select Export. Format of the export file can be either M3U or M3U8 (I think Miska prefers the latter for some reason); Save it to whatever convenient folder.

 

Back in HQP, just click the folder icon to load the playlist you saved. Those playlists are of course just pointers and not a duplication of the actual music files.

 

Hope the above very rough quick start helps in some way. There are other ways to use HQP (the whole library and full screen interface), but if you just want to get a feel for the SQ, then just start simple.

Have fun. Many HQ Player users here are more savvy with it than me, and I am sure they will be happy to guide you.

 

--Alex C.

 

An old post but very helpful for those just starting out with HQP. Thanks!!

[Hammer]

Hammer, go back to your DAC driver settings in HQPlayer settings menu and pick DoP (file -> settings -> SDM pack). It will ask you if you truly have a DoP acapbale dac, and answer yes. Then when you pick SDM(DSD) in the playback menu (the pulldown on the main screen, right under the big volume wheel..says either PCM or SDM) you have DSD playback via DoP.

 

Yes, a quirk in HQPlayer currently (hope it's fixed soon) is that the playlist goes away for many many reasons, the least of which is simply looking at your own settings. Due to the great sq I can live with that for awhile.

 

I tried the above settings, but I am just getting static. Interesting, however, that when I play the file, my DAC (Auralic Vega) will say DSD64. So, it seems like it recognizes the DoP header, but alas just static. Are there any other settings I can try? I am using the ASIO driver in Windows Server 2012R2.

 

Thanks, Hammer.

[Hammer]

Hi Hammer,

 

What settings did you use to get the tears?

 

Just the default settings at this point, but the sound quality is noticeably much improved for redbook files. It would be awesome if down the road, we can change settings on the fly to hear the differences real time. I can sort of do that on my DAC, but fumbling through the menus to get to the filter settings always takes longer than I want.

[tranz]

Well Miska, I have to say, amazing job on the filters. You wrote all the code for those?

 

Just using poly-sinc, NS9, 24 bits setting, 192Khz PCM and playing back a ripped CD spanked Amarra which until now was my go to baseline for RBCD.

 

What would be ideal from a consumer perspective would be DAC and Receiver manufacturers licensing your logic. "Powered by HQP" on the box. Even software playback companies like SonicStudio, JRiver, Apple should be licensing your algorithms.

 

I have not heard better, period!

 

The library management and user-friendly access and playback is the bit that is far from ideal, but why would you even want to compete with iTunes and their iPad Remote app. Combine iTunes, or a DLNA compatible version to run on a NAS with MinimServer, with your algorithms and everyone else can pack their bags.

 

Cheers

[fordski]

Well Miska, I have to say, amazing job on the filters. You wrote all the code for those?

 

 

The library management and user-friendly access and playback is the bit that is far from ideal, but why would you even want to compete with iTunes and their iPad Remote app. Combine iTunes, or a DLNA compatible version to run on a NAS with MinimServer, with your algorithms and everyone else can pack their bags.

 

Cheers

 

Tranz

 

Geoff Armstrong has done a lot of work creating scripts to actually allow the use of iTunes to feed music to HQPlayer. Check out his thread here: Play iTunes through HQPlayer

 

I've just tried it out today and it works as advertised allowing you to use iTunes, your iPad remote and play your music through HQPlayer. His last post in the thread has the latest script which also allows the use of other formats such as Apple Lossless.

 

Suggest you check it out.

[tranz]

Tranz

 

Geoff Armstrong has done a lot of work creating scripts to actually allow the use of iTunes to feed music to HQPlayer. Check out his thread here: Play iTunes through HQPlayer

 

I've just tried it out today and it works as advertised allowing you to use iTunes, your iPad remote and play your music through HQPlayer. His last post in the thread has the latest script which also allows the use of other formats such as Apple Lossless.

 

Suggest you check it out.

 

Hi fordski,

 

Thanks for letting me know. That sounds great, I will check it out.

 

Cheers

[EuroDriver]

On 3.7.0 it appeared due to CPU load peak causing drop-outs. Now it certainly shouldn't happen. If you still get clicks, something else is causing drop-outs or you get clipping (IOW, have volume set higher than -2 dBFS). On Windows or Linux, try increasing buffer time or lowering volume. On Mac OS X you can only adjust volume, buffer time is decided by CoreAudio.

 

I know from measurements that I'm not getting any clicks...

 

Miska,

 

Thanks for this, I will upgrade to 3.7.1 soonest

[bogi]

That does not provide bit perfect playback for PCM. Tried with bit rate set to default, 16, and 24 and those specific test files.

 

Hello tranz, how did you come to that conclusion? Could you please explain it?

[tranz]

Setting max to -3 dB is good way to avoid accidentally hitting limiter. -60 dB is very quiet usually, almost like mute, so if smaller adjustment range is suitable, then bringing up the minimum volume to -40 dB or something makes adjustment with mouse easier because the range is smaller.

 

Another way I use sometimes when preamp or integrated amp is used is to set max 0 and min to -6. Then the recommended -3 dBFS setting is 12 o'clock position of the knob and the steps are large.

 

 

 

Volume control for SDM can be enabled by unchecking the "DirectSDM" in DSDIFF/DSF Settings -dialog. Then also DSD content and PCM-to-SDM has volume control, but by default the direct mode is enabled to keep DSD truly direct from source to DAC.

 

 

Sorry necro-post, but Miska am I reading this correctly that there is a Limiter inbuilt?

Which one, as that can really mess with the sound?

 

I am hesitant to use any digital volume control, so checking DirectDSM will avoid this for DSD, but will setting both Min and Max to 0dB really disable it, i.e. optimal setting, for PCM?

 

After listening for a while it started getting a bit fatiguing, like a too hot signal. Perhaps related to the volume setting?

 

Thanks

[bibo01]

...

The library management and user-friendly access and playback is the bit that is far from ideal, but why would you even want to compete with iTunes and their iPad Remote app. Combine iTunes, or a DLNA compatible version to run on a NAS with MinimServer, with your algorithms and everyone else can pack their bags.

 

Cheers

For the highlighted mode you need HQPlayer Embedded (on Linux only) and have a chain like MinimServer->Rygel->HQP Embedded. Your control point would be BubbleUPnP.

I use it successfully with AudioLinux as OS.

[bogi]

I am hesitant to use any digital volume control, so checking DirectDSM will avoid this for DSD, but will setting both Min and Max to 0dB really disable it, i.e. optimal setting, for PCM?

 

After listening for a while it started getting a bit fatiguing, like a too hot signal. Perhaps related to the volume setting?

 

Thanks

 

Upsampling means interpolation and both theoretically and practically it can lead to increased signal peaks. Therefore limitation can occur also in the case when it did not occur without upsampling on the same volume level. That's the reason of Miska's recommendation NOT to use volume setting 0 if digital filters are in action.

 

Your fatiguing experience could be caused by signal limitation when you were using volume 0. You can check it by switching to full screen mode (the most right button in toolbar), then on left upper part of screen you will find the Limited counter.

 

For me the sound isn't subjectively the same with different volume settings in HQPlayer (and compensating that on preamp or headphone amp). It has technical reasons too. Lower level PCM signal is coded by smaller amount of bits, each 6 dB step causes loss of 1 bit. That means the range 0 to -6 dB can be coded by the same number of different digital values than the remaining range from -6 dB to max. attenuation. It does not occur how I described if for example DAC uses 24bit PCM resolution and you are playing 16bit recording. But analog volume regulation is typically still more suitable. Lowering volume digitally has negative impact to SNR (noise level does not depend on digital volume level). Further impacts are also possible, for example dithering works best with expected signal level range.

 

So more reasons exist why adjusting volume digitally and then compensating it in analog domain may lead to slightly different sound. You have to find what is best for you. 0dB with digital processing enabled is not the best because of possible limitation, that's clear. But lowering too much is also not ideal.

[tranz]

Hello tranz, how did you come to that conclusion? Could you please explain it?

 

Hi bogi,

 

On my DAC it provides a confirmation using specific test files whether a combination of bit rate and frequency are received bit perfectly by the DAC. I first check to get a baseline in setup in my current software Amarra, and then using the same files with HQP. No setting for PCM I tried provided bit perfect confirmation in HQP.

 

Cheers

[tranz]

For the highlighted mode you need HQPlayer Embedded (on Linux only) and have a chain like MinimServer->Rygel->HQP Embedded. Your control point would be BubbleUPnP.

I use it successfully with AudioLinux as OS.

 

Thank you. I saw a few older posts whereby the conclusion was that a NAS itself would not be powerful enough. I wonder if that is still the case if staying within PCM only.

 

So you have a NAS running Minim and another box running linux HQP? Are you then able to send the HQP stream output to an UPnP Renderer?

[tranz]

Upsampling means interpolation and both theoretically and practically it can lead to increased signal peaks. Therefore limitation can occur also in the case when it did not occur without upsampling on the same volume level. That's the reason of Miska's recommendation NOT to use volume setting 0 if digital filters are in action.

 

Your fatiguing experience could be caused by signal limitation when you were using volume 0. You can check it by switching to full screen mode (the most right button in toolbar), then on left upper part of screen you will find the Limited counter.

 

For me the sound isn't subjectively the same with different volume settings in HQPlayer (and compensating that on preamp or headphone amp). It has technical reasons too. Lower level PCM signal is coded by smaller amount of bits, each 6 dB step causes loss of 1 bit. That means the range 0 to -6 dB can be coded by the same number of different digital values than the remaining range from -6 dB to max. attenuation. It does not occur how I described if for example DAC uses 24bit PCM resolution and you are playing 16bit recording. But analog volume regulation is typically still more suitable. Lowering volume digitally has negative impact to SNR (noise level does not depend on digital volume level). Further impacts are also possible, for example dithering works best with expected signal level range.

 

So more reasons exist why adjusting volume digitally and then compensating it in analog domain may lead to slightly different sound. You have to find what is best for you. 0dB with digital processing enabled is not the best because of possible limitation, that's clear. But lowering too much is also not ideal.

 

Thanks bogi,

 

It really is a logical cause as it reminded me of that 'hot' sound on an old DAC where the dip switch had to be set to a lower voltage to match the pre-amp. The initial holy shite this sounds amazeballs slowly turned into wanting to revert back to Amarra for a soothing of the ears. On female vocals it was especially noticable as sibilance and shrillness set in.

 

I did not know about the full screen limiter counter. Thanks, as I can now experiment with volume settings a bit.

 

Cheers