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dasdoing

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So, 20 months and 26 pages later :

Debate not ended :p

I've explained somewhere in this thread why the discussed issue most probably is a non-issue, or why the limiter if audible is actualy benificial.
I also explained how to end the debate, and it's not hard and something the members here love to do: ABX some tracks where the limiter hits at 0 dBFS.
it's sad to waste headroom, and shouldn't be done on a theoretic assumption
 

daftcombo

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I've explained somewhere in this thread why the discussed issue most probably is a non-issue, or why the limiter if audible is actualy benificial.

When you have an accident, a safety belt is better than none. But the best is to try not to have an accident.
 

dasdoing

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When you have an accident, a safety belt is better than none. But the best is to try not to have an accident.

1) I strongly doubt that a single sample hitting the limiter at 0dBFS is audible; which is the situation probably 99% of the time.
2) when we have multiple sequential samples hitting 0dBFS it will cause intersample peaks way higher than 0dBFS. an MP3 will probably cause a "pop". Not sure if the limiter could avoid it.
the thing is, avoiding those accidents means going way lower than minus 0,2 dB. And than you have to ask how deep.
 

daftcombo

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the thing is, avoiding those accidents means going way lower than minus 0,2 dB. And than you have to ask how deep.

What about the answer just here?
Ages ago I measured a load of my tracks that I listen to in Orban Loudness Meter, and I determined that a negative preamp of -2dB would cover virtually every intersample over, not all but most, and -3dB would have covered all of them. So I run -3dB on my headphone setup as there's a lot of headroom available, and I run -2dB on my speaker setup. (I do run that in addition to the negative preamp required by the EQ, as I guess you can't be sure where in the frequency response the intersample overs are gonna occur.)

 

Jimbob54

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What about the answer just here?


But, let's just say you like your bass and add an 8db bass shelf, so - 8db preamp for eq plus another - 3db for overs to avoid the limiter per Robbo's post

-10 or 11 dB additional output is a heck of an ask for some headphone amps on some less sensitive headphones.
 

Mnyb

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My main takeaway from this tread is that , with today’s equipment it’s pointless to argue about how to preserve “bit perfect” it literally is not a goal for anyone anymore, if it ever was ? ( unless you test stuff ). More about what’s a sensible approach when you EQ and with a modern DAC throwing away some 10dB or more as headroom is not problematic.
Another takeaway is that even if you do some samplerate conversations etc they still have no audible impact even if the converter is just good not even sota but ok .
So your ok setting a common samplerate for everything and route trough EAPO and to your headphones and life is ok :) I’m convinced that I do some things ever so slightly wrong compared to perfection in my setup , but it still does not matter .

So I’ve learnt not to obsess over small things to much :) but starting to use the pc as an audio toolbox.
 

dasdoing

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What about the answer just here?



as I said, I don't think it makes sense to throw away headroom for a probably null effect.
if you guys still can't live with the remote possibility it beeing audible, I repeat: ABX it at least
 

daftcombo

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as I said, I don't think it makes sense to throw away headroom for a probably null effect.
if you guys still can't live with the remote possibility it beeing audible, I repeat: ABX it at least

It would be interesting to ABX the need for useless headroom too!
 

krabapple

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My main takeaway from this tread is that , with today’s equipment it’s pointless to argue about how to preserve “bit perfect” it literally is not a goal for anyone anymore, if it ever was ?

It's definitely a goal for me as all digital signals are sent 'raw' to my AVR. Some of those signals are Dolby Digital and DTS to be decoded in the AVR . If they aren't sent 'bit perfect' the result is white noise.

Bit perfect digital out is very easily achieved using WASAPI. All the EQ is then done in the AVR. What a huge load of... fuss, this thread is.
 

Mnyb

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It's definitely a goal for me as all digital signals are sent 'raw' to my AVR. Some of those signals are Dolby Digital and DTS to be decoded in the AVR . If they aren't sent 'bit perfect' the result is white noise.

Bit perfect digital out is very easily achieved using WASAPI. All the EQ is then done in the AVR. What a huge load of... fuss, this thread is.
A yes forgot about those I have some dts tracks :) but not many , I use the my pc for headphones not HT so forgot this use case . I’ve done they in the past with my music server .
 

Solano

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  • I know that I could also set an EQ in foobar but I don't want to change EQ every time I change between headphones and speakers.
if you setup eAPO like this you don't need to change manually:
1665271477200.png



Defined Output A
A Setting
A Setting
A Setting

Defined Output B
B Setting
B Setting
B Setting

For me it works perfectly on the fly.
 

daftcombo

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if you setup eAPO like this you don't need to change manually:
View attachment 236085


Defined Output A
A Setting
A Setting
A Setting

Defined Output B
B Setting
B Setting
B Setting

For me it works perfectly on the fly.
Don't you switch between outputs manually in PEACE? How does APO know?
 

Offler

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I've explained somewhere in this thread why the discussed issue most probably is a non-issue, or why the limiter if audible is actualy benificial.
I also explained how to end the debate, and it's not hard and something the members here love to do: ABX some tracks where the limiter hits at 0 dBFS.
it's sad to waste headroom, and shouldn't be done on a theoretic assumption

I did few tests on my system based on the initial assumption
a) The assumption was confirmed used FFT measuring.
b) I did try to play two signals at 18KHz and 19.5KHz at -0.14dB for IMD measuring and I confirmed that I have to sacrifice even more headroom. In this case the effect even worse and was very audible.
c) I tried the assumption on an extreme metal industrial music. The VA meter is always at 0 no matter what. And I am afraid that effect was audible. On less extreme sorts of music, the percussions were sounding different.

1. I apply the solution with EAPO to non-critical listening, where windows audio stack cannot be bypassed = mostly gaming.
2. For critical listening either CDs, FLACs, DVDs, BluRays i always use Exclusive mode where is no sacrifice to the headroom.

Also i checked most of my CD>FLAC conversions with Dynamic Range Meter. Some of the tracks were obviously artificially limited at -0.12dB - sometimes whole albums were mastered like that.
 

pvehling

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When using ROON set to ASIO with my E50 (with the ASIO driver installed of course), I'm still able to control volume from Windows and also DS audio is able to be played at the same time. I recall when using ASIO on my old RME Babyface FS in Roon, the main windows mixer was NOT able to work and DS audio didn't come through either. Is this by design of the ASIO driver for the Topping DAC? I'd prefer to have a clean ASIO signal over USB without any windows crap injecting itself into it..
 

shoto

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The Windows volume control still works in WASAPI or ASIO mode... how come?

Asio does not mute the sounds from other apps either.

^ I learned this is because of DAC using hardware volume control and not software control
 
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