HQPlayer Impressions and Settings Rolling Thread
Feb 23, 2024 at 7:55 PM Post #676 of 1,524
@jlaako on Desktop version 5 (for MAC) where can I find the ratio table that shows me which filter settings are better for 1x vs NX ?

When I click on preferences and then output - then go to the filter dropdown - there is no (integer) information displayed - so I am often left guessing which filter is best

P.S. I really love your product. Well done! Fantastic.
 
Last edited:
Feb 23, 2024 at 8:03 PM Post #677 of 1,524
I have what probably is a really stupid question.

I noticed that my version of HQPlayer is not the newest one and I want to update it to the newer version, my question is if it's a simple as just downloading the file and opening the .exe and just let it be?. I just want to make sure I'm not going to lose my license of HQP.(I'm talking here about the Windows Desktop version of HQP btw)
 
Feb 23, 2024 at 8:05 PM Post #678 of 1,524
I have what probably is a really stupid question.

I noticed that my version of HQPlayer is not the newest one and I want to update it to the newer version, my question is if it's a simple as just downloading the file and opening the .exe and just let it be?. I just want to make sure I'm not going to lose my license of HQP.(I'm talking here about the Windows Desktop version of HQP btw)
Yep just run the installer and it will overwrite the previous version keeping your license intact
 
Feb 23, 2024 at 8:12 PM Post #679 of 1,524
Feb 23, 2024 at 8:24 PM Post #681 of 1,524
I noticed that my version of HQPlayer is not the newest one and I want to update it to the newer version, my question is if it's a simple as just downloading the file and opening the .exe and just let it be?. I just want to make sure I'm not going to lose my license of HQP.(I'm talking here about the Windows Desktop version of HQP btw)
On Windows it is better first to uninstall HQPlayer in usual way from Settings and then to install a new version. Uninstallation is matter of few seconds. License is not touched with this, don't worry.

In 10 years I never needed to re-enter HQPlayer license. Anyway, store your license files at least at two different physical medias to have a reliable backup for the case of serious troubles with your computer.
 
Feb 23, 2024 at 8:38 PM Post #682 of 1,524
I'm interested in this as when recommending 20 bit depth, Jussi also provides pictures of the noise floor. But bit depth affects also other things than noise floor: it affects
Minimum dB step difference (quantization rounding error) (source: https://en.wikipedia.org/wiki/Audio_bit_depth). While with 20 bits the step is 0.000116 dB, with 24 bits is 0.00000871 dB, ie. over 10x improvement.

Correctly noise-shaped 20-bits gives you more dynamic range and linearity than 24-bit TPDF, but without the distortions of 24-bit.

In both cases, analog output is limited by reality of analog domain, since noise floor is defined by the DAC's analog components, not by the digital noise floor between these two cases which is anyway well below the analog one. But if you use something else than what I've recommended, the R2R non-linearity pops up as distortion.

I've shared number of measurements about this... If you use more bits, you add distortion, but you don't win anything in terms of dynamic range (since it's limited by Johson-Nyquist noise).
 
Last edited:
Feb 23, 2024 at 9:11 PM Post #683 of 1,524
- You have a device with a maximum output of 3V
- You need to output exactly 2V
- This can be accurately represented by a 2-bit system, because two bits can be '11' (100%/3V), '10' (66%/2V), '01' (33%/1V) or '00' (0%/0V). So you just choose '10' and you're good.
- If you now wanted to get this output with a 1-bit system, you can't do it accurately. A 1-bit system can only be '1' (100%/3V) or '0' (0%/0V). And so picking the closest value of '1' (3V), you now have a quantization error of 1V, the difference between the value you intended and the actual value.

1-bit system can give you about 1.9V absolutely perfectly though (mathematically) - and without the linearity errors you would get with 2-bit system. Add a bit of analog gain and you are at 3V output with much better linearity.

In addition, your 1-bit output can be any word length you desire.
 
Feb 23, 2024 at 9:29 PM Post #684 of 1,524
Thank you for that explanation. That makes it seem like there is no benefit to DSD over PCM in HQPlayer, because of all that potential error. What is the benefit of DSD over PCM? Is there any benefit of using DSD instead of PCM on a NOS DAC like a Holo Spring/May?

It all boils down the the actual D/A conversion. For example with Holo Audio DACs, the DSD conversion (with data from a good modulator) is notably better than what you get with PCM inputs.

Issue with PCM is combination of resolution and settling time to within ½LSB. You need high as possible rate to eliminate images from the output (distortion), but when you keep increasing sampling rate, the actual conversion stage has increasing trouble to settle withing specified output voltage within fraction of the sample period (which is getting shorter as function of sampling rate). In addition, it is physically impossible to create accurate enough resistors to gain 24-bit resolution using R2R solution. And even if you would be able to do so, thermal noise of those resistors in room temperature would overwhelm that and it would become moot. 0.01C change in room temperature would spoil that DAC completely.

So the solution is to reduce output word length as function of sampling rate and employ digital domain noise-shaping to maintain dynamic range in wanted bandwidth. This way the linearity is maintained, or even improved while the images get reduced too. So both Y (amplitude) and X (frequecy) axis improve. This is what delta-sigma (DSD) DACs do.

Ultimately you have just switching between two states (like class-D amplifiers too), at high rate. You only need to settle within ½ of the reference voltage. With class-D amplifiers we are talking about switching rates around 600 kHz. With DSD, we are talking about switching rates in tens of MHz.

Analog reconstruction filter is then used to remove the ultrasonic noise and reconstruct the final analog output signal.
 
Feb 23, 2024 at 9:33 PM Post #685 of 1,524
Halloh.

Welcome to the crazies of head fi. @jlaako
 
Last edited:
Feb 23, 2024 at 9:41 PM Post #686 of 1,524
It does not make sense to digitally strongly attenuate and then to amplify, because as I explained above with stronger digital attenuation you lose digital resolution.

As an example. ESS DAC chip has self-noise / output impedance equivalent of 600 ohm resistor. This is the ultimate analog limitation. Your digital domain noise floor is well below of that, no matter how your set the digital volume control.

Now if you have analog volume control with 10 kOhm potentiometer, that will be the one to spoil your SNR by large factor, not the digital volume control...

If you take let's say ASDM7EC-ul at DSD256, it'll will beat your analog domain by large margin...
 
Feb 23, 2024 at 9:47 PM Post #687 of 1,524
@jlaako on Desktop version 5 (for MAC) where can I find the ratio table that shows me which filter settings are better for 1x vs NX ?

Filters are not specific to 1x vs Nx. Except maybe "hires" ones are more geared towards Nx use.

If you prefer halfband-filters, those are better for Nx than for 1x. Since apodizing errors are more prominent to 44.1/48k sources than hires.

When I click on preferences and then output - then go to the filter dropdown - there is no (integer) information displayed - so I am often left guessing which filter is best

I hope this is where the table in the manual becomes handy...

On Windows and Linux, you can find manual from HQPlayer-group of Start-menu. On macOS, you can just drag the manual anywhere you like from the DMG, for example on your desktop.
 
Feb 23, 2024 at 10:13 PM Post #688 of 1,524
It all boils down the the actual D/A conversion. For example with Holo Audio DACs, the DSD conversion (with data from a good modulator) is notably better than what you get with PCM inputs.

Issue with PCM is combination of resolution and settling time to within ½LSB. You need high as possible rate to eliminate images from the output (distortion), but when you keep increasing sampling rate, the actual conversion stage has increasing trouble to settle withing specified output voltage within fraction of the sample period (which is getting shorter as function of sampling rate). In addition, it is physically impossible to create accurate enough resistors to gain 24-bit resolution using R2R solution. And even if you would be able to do so, thermal noise of those resistors in room temperature would overwhelm that and it would become moot. 0.01C change in room temperature would spoil that DAC completely.

So the solution is to reduce output word length as function of sampling rate and employ digital domain noise-shaping to maintain dynamic range in wanted bandwidth. This way the linearity is maintained, or even improved while the images get reduced too. So both Y (amplitude) and X (frequecy) axis improve. This is what delta-sigma (DSD) DACs do.

Ultimately you have just switching between two states (like class-D amplifiers too), at high rate. You only need to settle within ½ of the reference voltage. With class-D amplifiers we are talking about switching rates around 600 kHz. With DSD, we are talking about switching rates in tens of MHz.

Analog reconstruction filter is then used to remove the ultrasonic noise and reconstruct the final analog output signal.
Thank you for that explanation. I love your product. Do you think you will add the option of Tidal login to HQPlayer Client?
 
Feb 24, 2024 at 1:12 AM Post #689 of 1,524
Yep just run the installer and it will overwrite the previous version keeping your license intact
That's correct, however for HQ player desktop on Windows I believe it is recommended to uninstall the prior version before installing the update. Settings will be preserved even when doing so.
 
Last edited:
Feb 24, 2024 at 4:13 AM Post #690 of 1,524
As an example. ESS DAC chip has self-noise / output impedance equivalent of 600 ohm resistor. This is the ultimate analog limitation. Your digital domain noise floor is well below of that, no matter how your set the digital volume control.

Now if you have analog volume control with 10 kOhm potentiometer, that will be the one to spoil your SNR by large factor, not the digital volume control...

If you take let's say ASDM7EC-ul at DSD256, it'll will beat your analog domain by large margin...

Hi Jussi, my thinking is based on this:

Lets have a 24bit recording and DAC with 20 bits of analog domain resolution. Lets have a chain without analog preamp, so using HQPlayer volume control as preamp (preatt) with power amplifier directly connected to DAC. So no analog preamp with volume pot is used.
Now I will attenuate to -60 dB, what approx. means shifting the audio data about 10 bits right, with adding 10 zeros at beginning.

Both HQPlayer oversampling filters and modulator increase dynamic range. It is hard to know for me how much exactly, but since it is mostly enough to use -3dB to -6 dB of attenuation to avoid signal limitation, for simplicity I will count with 1 bit of improvement over 10 lost bits in my thought example.

Valid audio data of original 24 bit recording:
rrrrrrrr rrrrrrrr rrrrrrrr
After 60 dB of attenuation:
00000000 00rrrrrr rrrrrrrr rrrrrrrr rr (10 zeros at beginning)
Valid audio data after improving about 1 bit with upsamling/modulation (simplified for the purpose):
00000000 0rrrrrrr rrrrrrrr rrrrrrrr rr (9 zeros at beginning)
Now when we limit the output to 20 bits of DAC resolution, we get only the upper part, containing only 11 bits of valid audio data (max. 66 dB of dynamic range):
00000000 0rrrrrrr rrrr
And the rest of valid audio content appears now lost below DAC analolg noise floor.

And now, after such a strong attenuation, power amp is amplifying analog signal, which contains only part of original audio information, since low level details were lost below DAC noise floor.

Now let think about the opposite situation - small digital attenuation to avoid limitation in HQPlayer and rest required attenuation done by analog preamp.
The goal is to fill upper 20 digital bits with valid audio data so DAC will use its full 20 bits of resolution to create analog output. Then let the required attenuation to be done in analog domain.
If we would attenuate only about 6 dB (instead of 60) digitally and if we would be getting the same 1 bit of dynamic range back by means of upsampling and modulation, we get:
Original recording
rrrrrrrr rrrrrrrr rrrrrrrr
After 6 dB of attenuation:
0rrrrrrr rrrrrrrr rrrrrrrr r (1 zero at beginning)
After improving about 1 bit with upsamling/modulation (simplified):
rrrrrrrr rrrrrrrr rrrrrrrr r (no zeros at beginning)
Now when we limit the output to 20 bits of DAC resolution, we get 20 bits of valid audio data, so 9 bits of audio information more than in the 1st case.
rrrrrrrr rrrrrrrr rrrr

Now if we would further attenuate with analog preamp, we attenuate analog audio content containing much more low level details (maximum what DAC is able to provide). So the signal at analog preamp output will contain also lower level details, which were missing at DAC output in the 1st case, but attenuated. Of course, analog preamps don't behave ideally, so part of that low level detail could be shifted below preamp noise floor. But we cannot generalize how much - it depends on preamp implementtaion.

My example is simplified to be illustrative but you get my point. What's wrong in my thinking?
 
Last edited:

Users who are viewing this thread

Back
Top