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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
 

Hov

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I suppose I'll always implement at least the 0.2. If I've got the headroom I may "feel better" with it even lower but I understand your point @dasdoing
 

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!
 
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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.
 

formdissolve

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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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julitoole

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Hey guys,
I have some questions if somebody cares to shed some light in my brain of darkness.
I have a dx5 and apo and peace installed. The device is capable of 32 bit 784khz. In the Audio device properties I set the device to 32 bit and 382khz I see no option to set it to 32 bit 784khz.
But if I understood right that is not at all the way I should set it as it automaticaly upsamples?
If I understood right I should find a middleground for example 96khz as most of music doesnt exceed that samplerate?
Same thing with bitrate. Should I set it to 32 bits because my device is capable of it? or is 24 bits better or worse?

Then another question I read something about a dynamic sample rate? This would come in handy as I have to constantly change the sample rate for a game that refuses to play sound if it is set to anything else than 48khz. So how do I set that dynamic samplerate?

Then I don´t get dither, for me the explanation for doenst make sense to me. As I understood dither adds noise to the track so recreation in DAC´s or similar doesnt mistake a stray fragment as a sound it should recreate in analog. But I thought this has to be added in the track as it was created. But I can set dither in my dx5 what does it do? I didnt hear any difference so this seems negligible to me?

Im not the smartest bulb here in the forum so please be kind to me xD.

Best regards Julius
 

Robbo99999

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Hey guys,
I have some questions if somebody cares to shed some light in my brain of darkness.
I have a dx5 and apo and peace installed. The device is capable of 32 bit 784khz. In the Audio device properties I set the device to 32 bit and 382khz I see no option to set it to 32 bit 784khz.
But if I understood right that is not at all the way I should set it as it automaticaly upsamples?
If I understood right I should find a middleground for example 96khz as most of music doesnt exceed that samplerate?
Same thing with bitrate. Should I set it to 32 bits because my device is capable of it? or is 24 bits better or worse?

Then another question I read something about a dynamic sample rate? This would come in handy as I have to constantly change the sample rate for a game that refuses to play sound if it is set to anything else than 48khz. So how do I set that dynamic samplerate?

Then I don´t get dither, for me the explanation for doenst make sense to me. As I understood dither adds noise to the track so recreation in DAC´s or similar doesnt mistake a stray fragment as a sound it should recreate in analog. But I thought this has to be added in the track as it was created. But I can set dither in my dx5 what does it do? I didnt hear any difference so this seems negligible to me?

Im not the smartest bulb here in the forum so please be kind to me xD.

Best regards Julius
Hi, I'll let others respond to your other questions, but re Sample Rate & bit rate, just set the sample rate to whatever source you're gonna be using. So if your music is recorded at 44kHz, then set it to that when listening to music......but then when you're gaming probably set it to 48kHz as most games use 48kHz (also for movies) - so just set sample rate to match your source. For bit rate just set the highest - 32bit, because if you are using digital volume control then it more accurately preserves the available "volume steppings" within the original recording when running at low digital software controlled volumes. There is no harm at running it at 32bit, only positives.
 

Roen

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So here's something I'm sure people in this thread would find interesting: I found a way to completely remove CAudioLimiter from the audio path!

What I realized is that CAudioLimiter is an APO like any other, and like all APOs, it has its own registered COM class with its own CLSID. This blog post has the CLSID - it's {D69E0717-DD4B-4B25-997A-DA813833B8AC}. We can deliberately mess up that class registration so that the Windows audio engine cannot find it anymore. Luckily, if the Windows audio engine can't load that APO, then audio still works - it just bypasses the missing APO!

Here's the procedure:

  1. In the Registry Editor, navigate to HKEY_LOCAL_MACHINE\SOFTWARE\Classes\CLSID\{D69E0717-DD4B-4B25-997A-DA813833B8AC}.
  2. You're gonna have to mess with the permissions to be able to make the required changes. Change the owner of the InProcServer32 subkey to Administrators, then give Administrators Full control.
  3. Make any change that would prevent the class from being loaded. For example, change the (default) value to add a "DISABLED" prefix so that the path to audioeng.dll is broken.
  4. Restart the Windows Audio service (audiosrv).

After I did this, audio still worked fine but I was unable to trigger the limiter - the audio pipeline goes all the way up to 0.00 dBFS without any additional distortion. I think it should be truly bit-perfect now (in 24-bit at least - in 16-bit there is still dithering), but I haven't confirmed it.
You would still need the -4 dB to avoid inter-sample overs?
 
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