HQ Player

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6 hours ago, bogi said:

I would appreciate if you could upload a snippet (for example 30s) of such a track.

Sure. I didn't make a note of those tracks. But I'll use bs2b from time to time in the near future and come back to you as soon as I've spotted a track. If memeory serves me well it happend with some modern pop/rock tracks, so I'll start there.

 

6 hours ago, bogi said:

I could listen to SPL Phonitor mini some years ago and I quite enjoyed its rather darker presentation as well as its crossfeed functionality.

Don't know the Mini but the new Phonitor One is not really dark. It's powerful, beefy but overall well balanced and highly detailed I would say. However, I don't listen to headphones a lot, I'm a speaker guy. Therefore I have limited experience with headphone amps (and heaphones) so take my impression of its sound with a grain of salt.

 

 

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19 hours ago, KenMoreira said:

where can i find a guide for building an EQ in in HQplayer . Similar to that of say APO / Peace GUI?
I'd like to be able to pick my frequency range , volume of that, and of course the phase.
and he negative value pre to prevent clipping. 

Not sure what exactly you are asking but easiest way to apply EQ in HQP ist to create a TXT-File and load it into the matrix pipeline (one TXT for each channel, so twice the same TXT-File for channel 1 and channel 2 for stereo).

Syntax is the same (naming for some filters not exactly the same but similar) as in APO EQ.

 

Syntax for a filter set looks like this (numerical value = XX):

Filter: ON [FILTER-TYPE] Fc XX Hz Gain +X.XX dB Q X.XX

 

Short names for Filter-Types:
PK = Peak
LS = LowShelf
HS = HighShelf
HP = HighPass (= LowCut)
LP = LowPass (= HighCut)
(and more - they are in HQPs manual)

 

Q or S for filter bandwidth

Fc for frequency

Gain (+/-) for gain in dB

Preamp for preamp (+/- XX dB)

Attached a screenshot of a random setting containing Preamp, peak and shelf filters.

 

Phase can not be adjusted. The EQ is minimum phase by design.

 

You can also create a filter set in REW and export it as TXT.

 

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1 hour ago, KenMoreira said:

Preamp: - 5.5 dB
Filter:  ON PK Fc 73 Hz Gain -1.4dB Q 0.9
Filter:  ON LS Fc 105 Hz Gain +5.5dB Q 0.71
Filter:  ON PK Fc 230 Hz Gain -1.7dB Q 1.0
Filter:  ON PK Fc 480 Hz Gain -2.0dB Q 5.0
Filter:  ON PK Fc 1000 Hz Gain -1.9dB Q 2.2
Filter:  ON PK Fc 1850Hz Gain +5.3dB Q 0.9
Filter:  ON PK Fc 2800Hz Gain -4.2dB Q 2.0
Filter:  ON PK Fc 4800Hz Gain -3.3dB Q 3.0
Filter:  ON HS Fc 10000Hz Gain +0.5dB Q 0.71 Is what I can up with.. But it crashed when i loaded it to channels 1 and 2 and saved .. 

You have to take care of formatting and syntax.

 

Several errors in your setting:

Preamp - delete blank space between "minus" and "5.5"

All rows - insert blank space between Gain value and "dB"

from Fc 1850 and up - insert blank space before "Hz"

 

So like this:

Preamp: -5.5 dB
Filter: ON PK Fc 73 Hz Gain -1.4 dB Q 0.9
Filter: ON LS Fc 105 Hz Gain +5.5 dB Q 0.71
Filter: ON PK Fc 230 Hz Gain -1.7 dB Q 1.0
Filter: ON PK Fc 480 Hz Gain -2.0 dB Q 5.0
Filter: ON PK Fc 1000 Hz Gain -1.9 dB Q 2.2
Filter: ON PK Fc 1850 Hz Gain +5.3 dB Q 0.9
Filter: ON PK Fc 2800 Hz Gain -4.2 dB Q 2.0
Filter: ON PK Fc 4800 Hz Gain -3.3 dB Q 3.0

 

This is one of the reasons why I prefer to align all columns (by instering blank spaces) like this:

Ken formatted.txt

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3 hours ago, bogi said:

6 hours ago, Audionumber3 said:

I'm trying to find filters I can just leave it on..... Set and forget. But that's easier said than done...LOL.

That's probably not going to be the case. But you can accustom to a smaller collection of your favorite filters and you can try to remember for which kind of music/sound you liked which one and why.

Well, the HQP manual lists music-genres per filter. All the filters that are qualified for "any" music style are really good as "set it and forget it" filters. Great choice on behalf of Miska, for sure!

I for one use poly-sinc-gauss almost always (would love to use poly-sinc-gauss-halfband always, but there are too many albums in my collection that need an apodizing filter). I used to switch to poly-sinc-short-mp to slightly smooth out the top end with certain rock music. But since I love to switch things on the fly meanwhile I prefer to stay with poly-sinc-gauss and apply an EQ setting to get a similar effect. Rarely do I switch to poly-sinc-xtr-short lp (on headphones) or poly-sinc-xtr-short-mp. Very ralrely to a longer filter for classical music (sounds too good also with ps-gauss to bother).

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14 hours ago, Miska said:

It is still on of my go-to for old prog-rock. Pink Floyd's Meddle-album, especially San Tropez track was used a lot when working on that one! Not for smoothing top end, but for natural snappiness sound of the drums/percussions etc.

I'm sure that smoothing something has never been a design target of anything in HQPlayer 😂

But the side-effect of early roll-off, minimum-phase, short ringing and apodization can make a fuzzy top end a bit more silky/elegant.

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great updates in v5.2.0 👍

 

@Miska

If time permits - would it be possible to post a graph/plot of the transition band of the new ps-gauss-halband-s filter?

And maybe also of the ps-gauss-halfband?

Many thanks in advance!

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9 minutes ago, bogi said:

I would expect that all internal filters in DAC chips are not apodizing

most are. But depending on the DAC Chipset there might be an Apodizing Filter to select from the menu; for instance ES9038Pro chip provides this option.

While these built in filters are apodizing they often also show quite some passband ripple (which might have an detrimental audible effect or not).

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@Miska

A question re the gauss-hb and gauss-hb-s filters.

The manual says they are "(slightly) leaky around Nyquist but extremely high attenuation".

May I ask how much attentuation they provide?

The new gauss-hb-s is great! :-)

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2 hours ago, Miska said:

Actually the attenuation comment in manual regarding gauss-halfband-s is a copy-paste error I have already fixed for next release.

 

Both have very similar transition, but halfband-s trades attenuation to do same in shorter length. They reach full attenuation by 24.1 kHz which is where image of 20 kHz tone would be.

Many thanks, Miska - much appreciated!!

 

2 hours ago, Miska said:

"Only suitable for highest technical quality source materials". Which means it is not suitable for 95+% of RedBook content.

This morning I‘ve recorded the output of my Desktop-DAC with gauss-hb and gauss-hb-s.

With a Tascam DR100 🙄 - but, hey, that‘s all I have to digitally record analog line level input. Anyway.

I have a 44.1kHz/16 album that doesn‘t trigger the Apodizing Counter in HQP due to a steep high quality LPF but the actual musical content goes up to 22K (with quite some energy).

While I would find the images produced by the hb filter at least acceptable (images peak below -90dB) the hb-s filter shows pretty strong images. It‘s fine, though, with recordings that roll off gently and don‘t contain high levels above 20kHz.

So I would say the Apodizing Counter is a great guide if you use for instance ps-gauss-short but the use of gauss-hb-s sure requires further attention. With an appropriate recording it‘s really great!

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@Miska

Today I've played around with filters.

Transcoded some stuff (dirac impulses and music-cuts) through HQP Pro and found that poly-sinc-short (lp/mp) produces mirrored images.

Is ps-short not supposed to be apodizing?

44.1kHz/16 upsampled to 88.2kHz/24 (ps-short-lp):

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7 minutes ago, Miska said:

It is apodizing, but as name says, it is short (very short) slow roll-off.

ok, thanks!

But itsn't the main purpose of apodizing filters to surpress images?

I've compared ps-short to ps-gauss short and the latter produces even a little less images (due to higher attenuation I assume) although it is non-apodizing.

I am confused 😐

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38 minutes ago, Miska said:

 

No, it is to correct errors in the source data and replace the ADC decimation filter's impulse response with alternative one.

 

All filters are supposed to suppress images.

 

For example gauss-xla and gauss-xl both suppress images. But only former one is apodizing.

 

 

Yeah, it is not exactly black and white on borderline cases like these. I should actually mark gauss-short as apodizing as well. For some reason I have not... The two are quite similar - but different.

 

Makes snese - many thanks 👍

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9 hours ago, Miska said:

 

You see that's why I have not been putting numbers and graphs to the manual. But instead overall textual descriptions and suggestions. Some number, or a graph don't really give a good answer. Answer is more complex, it would be long series of mathematical formulas.

While I also find graphs enlightening quite often I am finding your way to describe the filters extremely helpful!

For example: if you look at the graphs of ps-gauss (white) and ps-xtr-short (blue) one could think ps-xtr-short reproduces more "space".

But due to the "design" of the filters infact ps-xtr-short clearly sounds more intimate - you really focus on the tones of each instrument. It sounds incredibly detailed. So "Timbre" is an accurate way to describe the characteristic of this filter (IMO). I love this filter when listening to headphones (with Rock/Pop/Electronic/Jazz/Accoustic)

 

Then there's of course also personal perception. Me for instance I am having a hard time to agree on the different descriptions for ps-gauss and ps-gauss-long. Looking at the frequency response plots I wouldn't expect that I could hear a difference between the two at all. The description in the manual states, "timbre" and "transients" for both and for the long version also "space".

Strangley I perceive it the other way around: I am finding the ps-gauss filter more "open" and "airy" than ps-gauss-long. Only plausible explanation I can think of is I am sensitive to the longer ringing in ps-gauss-long (which in itself has no "long" ringing of course, but somewhat longer than ps-gauss). Maybe that doesn't make sense at all, but that's what I am hearing.

 

*edit: added gauss vs gauss-long below for reference

 

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8 minutes ago, Miska said:

As I've said it is more like suggestion and rough guidance, rather than absolute truth how something would be perceived.

Yes, I know. And as I've said I am finding your descriptions extremely helpful.

I rather question my perception than your descriptions :-)

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12 hours ago, Miska said:

I'll add this to my TODO-list. I agree that the list is getting rather long to scroll around...

That would be super cool!

I would appreciate a kind of "favorites" selection that is on top of the list - but all the rest is still there below.

I've organized my matrix profiles like this - my list starts with duplicatates of my (7) most used settings and the entire list comes below. At least on iPhone/iPad this works really great.

 

Just a suggestion :-)

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40 minutes ago, EMINENT said:

 

+1 for preset tilts.

 

Btw, how do you normally do your tilts?

 

I use this method for headphones.

 

iir:type=hshelf;f=63;g=-0.66;q=0.50,iir:type=hshelf;f=632;g=-0.66;q=0.50,iir:type=hshelf;f=6324;g=-0.66;q=0.50

 

well, actually I only posted my matrix preset list as an corresponding example how I would appreciate to create a favorites SRC filter list.

 

But regarding tilts yours (green) is a little bit wavy. Not that it matters much, but for a really straight tilt (pink) you need 5 evenly cascaded 1st order shelf filters; you use only 3.

 

Meanwhile I always use these values for LowShelfs: 50Hz (x4=) 200Hz (x4=) 800Hz (x4 =) 3200Hz (x4=) 12800Hz.

 

For a -2db downward tilt your 3x -0.66dB would be 5x -0.4dB with my settings.

So like this (I use Text files):

Filter: ON HS Fc    50 Hz Gain -0.40 dB S 0.5
Filter: ON HS Fc   200 Hz Gain -0.40 dB S 0.5
Filter: ON HS Fc   800 Hz Gain -0.40 dB S 0.5
Filter: ON HS Fc  3200 Hz Gain -0.40 dB S 0.5
Filter: ON HS Fc 12800 Hz Gain -0.40 dB S 0.5*

 

 

* "S 0.5" = "q=0.50", so the same

 

 

 

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3 hours ago, bogi said:

 

Look here:

 

Hi bogi,

 

if you still use the script you can reduce the 10 doubled frequencies to 5 quadrupled frequencies for the 1st order LowShelfs. As posted above:

50Hz (x4=)

200Hz (x4=)

800Hz (x4 =)

3200Hz (x4=)

12800Hz

 

(and of course twice the gain value for each filter for the same outcome as the 10-point version)

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8 hours ago, bogi said:

 

I am still using more years old outcome of that script.
I will re-run it according to your recommendation.
Thanks!

your welcome!

While we're at it.

"Tilt" usually means the lower spectrum gets boosted and the upper spectrum gets attenuated (or vice versa).

Since (here) we only use HighShelfs with negative gain everything gets attenuated. So it makes sense to level match the tilts near the center frequency by adjusting the preamp gain. Center frequncy is actually 632Hz but for convenience I myself use 500Hz.

 

This way switching filters on the fly also doesn't trick our perception too much. When you only use attenuated HighShelfs things seem to sound dull at first when you switch to a stronger tilt. In truth everything gets somewhat quiter and that influences perception a lot. But when the tilts are level matched somewhere at the center of the spectrum you don't perceive a volume change but in fact what a tilt is supposed to do: linearly boosting the lower spectrum and attenuating the upper (or vice versa).

 

(instead of gain adjustment of the whole curve you could also make the 2 lower frequencies LowShelfs with corresponding positive gain instead of HighShelfs with negative gain. In this case - based on my curve points above - the curve tilts somewhere around 400Hz, which is also ok for a coherent listening experience when switching filters.)

 

Just for completeness...

 

 

 

 

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4 minutes ago, Kalpesh said:

what is the rational for those tilt things ?

Basically tilts are used to EQ.

But, and IMO this is a big "but": a linear tilt fully preserves the relation of the instruments (tones) to each other in music - it's the least intrusive way to make the whole piece "brighter" or "darker".

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Your approach is perfectly fine!
But there are many ways to skin a cat.

 

The tilts come into play when you think about "tone controls" (bass/treble).
It's just a certain way to EQ your music (in a parametric EQ, BTW) in a non-intrusive way.

 

7 hours ago, Kalpesh said:

thank you. I'm not sure I understand the rational though.

(...)

Say I'm annoyed by a bit too much treble (...) I then will lower my treble response but certainly not lift my bass response simultaneously.

Music is always about balance. When you lower the treble and apply the same gain values inverted for overall gain - then you have boosted bass (+ mids, depending on the cutoff frequency and slope of your "treble" filter).

Simplyfied speaking: when you lower the treble by a certain level (let's say -2dB) a corresponding tilt would lower the upper spectrum by the half of that level (-1dB) and boost the lower spectrum by a half that level (+1dB). The relation is the same.

But, of course, depending on the cutoff frequency and the slope of your treble filter the outcome may be pretty different.

if you want to adjust for instance "shoutiness" or "harshness" of a recording a tilt will not do it because it's not meant for corrective EQ.
As said above: Linear Tilts preserve the spectral balance of the recording; they are the most "transparent" way to make music "brighter" or "darker". If you don't like the sound or are more accustomed to other ways to EQ the music to your liking that's perfectly fine. There is nor wrong or right.

 

BTW: the room target curves - beside the optional bass boost - are more or less tilts [either linear (-1db/octave overall), almost linear (B&K), flat to 1kHz or 2kHz and after that linear -1db/octave towards 20kHz).

 

 

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2 hours ago, Kalpesh said:

I'd truly like to understand (...) why it's supposedly most transparent.

Actually it's already in my post above: a linear tilt preserves the spectral balance.

EQing always introduces some kind of coloration (how much of course depends on the shape/slope/steepness/gain of the filter); a linear tilt doesn't do that (at least to the smallest possible amount).

 

Downside is it's not felxible 😀 - it's really just: "make my music brighter / less bright" without altering it's character!

When I do listen to music I don't want to think about which house curve the mastering engineer was using or think about anything at all. When I feel a certain album sounds too bright I simply apply a moderate downward tilt on the fly (if it's still too bright I select a slightly steeper tilt). I don't even have to think about the cutoff frequency or whether I use a shelf or a bell filter to make it less bright. Really like a bass/treble tone control bulit into a good old HiFi amplifier (that often had broad Low- and HighShelfs centered somewhere around 1kHz).

Now, I also use more specific EQ settings. But basically I am happy only with tilts and P-Loudness settings.

 

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7 hours ago, StreamFidelity said:

Can someone explain this spectrogram to me? The Apod counter shoots up.

 

That's some really ugly upsampled 96kHz version. You can clearly see the mirrored spectrum.

I have a 48kHz version from Highresaudio which doesn't trigger HQPs Apodizing counter.

Just looked again at the site and the say:

"Please Note: We offer this album in its native sampling rate of 48 kHz, 24-bit. The provided 96 kHz version was up-sampled and offers no audible value!"

If you ask me this is also BS - to me it clearly looks like upsampled 44.1kHz, not 48k.

Should have read it before buying 😬 ... in this case I'd prefer the 44.1kHz/16bit version.

Maybe I'll buy it again in 44.1kHz/16bit because it's a great Album!

 

edit: just did that.

If someone wants a 44.1k version uspampled to 96k and downsampled to 48k - I have one for free 🤪

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On 10/31/2023 at 2:09 AM, Miska said:

I have also two HS02's now, but I have not tested those yet. The slightly annoying part is the need for external PSU and the question how well that is isolated from the output. Not huge fan of having one extra variable in the critical path...

Reading the specs of the HS02 the external PSU is only needed when the connected USB device (DAC) requires 5V - the HS02 itself is powered through USB from the host computer.

With Intonas 7055 series it's actually similar: there's an optional AUX power connection to feed the connected USB device if its current requirements exceeds 500mA. Up to 500mA no external power supply is required. Plus: the Intonas outgoing 5V voltage is linear.

Intona devices are made for industrial and medical usage and such - back then when I bought my first Intona I appreciated the fact Intona has nothing to do with audio 😁

 

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@Miska

what‘s the purpose (a possible use case) of converting parametric EQ to convolution EQ?

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11 hours ago, Miska said:

Another thing is possibility to convert it to a linear phase EQ filter.

Thanks Miska!

Great option!

What kind of linear filtering did you implement? Forward-Backward? What’s the delay (length)?