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Minimum Phase vs Linear Phase

hyperknot

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Right. The only thing to concern is you need headroom for software upsampling, so you need to use software volume control/management as well, and because software upsampling happens before entering the DAC, you cannot substitute it with the intersample headroom (if any) in the DAC. That means any hardware headroom within the DAC would be wasted.

Yes, I'm using a -4 db headroom in software, even with this I almost never use my preamp's volume over 1-2 o'clock, so loosing possible max volume isn't really a problem.

If the problem is loosing 1-2 bits of possible information, I'm not so concerned about it, I think there is still plenty if resolution in today's DACs. And in case of 44/16 material, I think I'm still way over the 16 bits in the source material.
 

Julf

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And an other naive question: since this is all happening in the digital domain, why is it not an option to remove all ultrasonics after upsampling? In Adobe Audition I can select an area on the FFT graph and just press delete and those frequencies are then gone. Wouldn't this solve the problem of ultrasonics?

At some point it has to be converted to analog, and the conversion itself does need a low-pass reconstruction filter - but if you have upsampled and filtered in the digital domain, you can use a DAC filter with a very mild slope and place the frequency high enough that it isn't an issue.
 

bennetng

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And an other naive question: since this is all happening in the digital domain, why is it not an option to remove all ultrasonics after upsampling? In Adobe Audition I can select an area on the FFT graph and just press delete and those frequencies are then gone. Wouldn't this solve the problem of ultrasonics?
Well, because Audition runs on a much more powerful x86-64 processor. Before someone shills on this thread for expensive software solutions, let's see how freeware can do:
https://www.audiosciencereview.com/...gh-of-a-sample-rate-is-enough.4037/post-95240
https://www.audiosciencereview.com/...le-review-measurements-also.11868/post-398661
 

hyperknot

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Well, because Audition runs on a much more powerful x86-64 processor. Before someone shills on this thread for expensive software solutions, let's see how freeware can do:
https://www.audiosciencereview.com/...gh-of-a-sample-rate-is-enough.4037/post-95240
https://www.audiosciencereview.com/...le-review-measurements-also.11868/post-398661

I don't think it takes that much processing power (it's a few seconds after hitting delete), and a Raspberry Pi 4 is defintiely quite powerful these days. I think I'm just not getting something in the theoretical level. What would happen if we'd use a linear filter + remove ultrasonics manually?
 

mansr

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I don't think it takes that much processing power (it's a few seconds after hitting delete), and a Raspberry Pi 4 is defintiely quite powerful these days. I think I'm just not getting something in the theoretical level. What would happen if we'd use a linear filter + remove ultrasonics manually?
That delete operation of yours is a (most likely linear phase) low-pass filter.
 

bennetng

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I don't think it takes that much processing power (it's a few seconds after hitting delete), and a Raspberry Pi 4 is defintiely quite powerful these days. I think I'm just not getting something in the theoretical level. What would happen if we'd use a linear filter + remove ultrasonics manually?
So do you see ultrasonic differences in Audition's FFT view if you use Audition's built-in upsampler with high quality settings? I suppose you don't need to delete any ultrasonic images manually, right?
 

hyperknot

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So do you see ultrasonic differences in Audition's FFT view if you use Audition's built-in upsampler with high quality settings? I suppose you don't need to delete any ultrasonic images manually, right?

You are right, I tried it with a sweep and it gave very clean output.

But then I'm confused, why do we see ultrasonic artifacts here? This is with a very high quality SoX resampling. Source
DFC.png
 

bennetng

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You are right, I tried it with a sweep and it gave very clean output.

But then I'm confused, why do we see ultrasonic artifacts here? This is with a very high quality SoX resampling. Source
DFC.png
The spikes are harmonic distortion and intermodulation distortion in analog domain. The rise of noise floor above 50kHz are caused by delta-sigma ADC-DAC chain.
 

hyperknot

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The spikes are harmonic distortion and intermediation distortion in analog domain. The rise of noise floor above 50kHz are caused by delta-sigma ADC-DAC chain.

Yes, I'm referring to the green spikes there. So you are saying that if we'd feed this DAC with a 192 kHz input, generated from 19 & 20 kHz sine wave, it'd also have the spikes?
 

Robbo99999

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Months ago I've listened to Linear Phase vs Minimum Phase filters on my SoundblasterX G6 via JDS Atom headphone amp and AKG K702 headphones, and I was quite sure at the time that I liked the option "Fast Roll Off - Linear Phase" more than any of the others...to me that filter brought out the details of intimate female vocals better...I could hear more character in intimate female vocals...that sounds very subjective and it is, but that's the best way I can describe it....I kept replaying a certain part of the song "It Could be Sweet" from Portishead whilst switching the filter, and that's where I noticed the difference.

Since that time I've read a lot here on these forums, and that does make me doubt my earlier conclusion that I could notice a difference, but I'm not gonna go retesting those filters, I'm gonna leave it on the one I thought I liked the best "Fast Roll Off - Linear Phase".
 

John Dyson

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Months ago I've listened to Linear Phase vs Minimum Phase filters on my SoundblasterX G6 via JDS Atom headphone amp and AKG K702 headphones, and I was quite sure at the time that I liked the option "Fast Roll Off - Linear Phase" more than any of the others...to me that filter brought out the details of intimate female vocals better...I could hear more character in intimate female vocals...that sounds very subjective and it is, but that's the best way I can describe it....I kept replaying a certain part of the song "It Could be Sweet" from Portishead whilst switching the filter, and that's where I noticed the difference.

Since that time I've read a lot here on these forums, and that does make me doubt my earlier conclusion that I could notice a difference, but I'm not gonna go retesting those filters, I'm gonna leave it on the one I thought I liked the best "Fast Roll Off - Linear Phase".

I want to reassure that a LONG not-linear-phase FIR filter with different delays vs frequency just might sound different than a linear-phase filter with the same delay for all frequencies. The problem with so many of the discussions is that they devolve into the audibility of phase, rather than recognizing that a long filter that doesn't have constant delay vs freq can have relatively large TIME differences because of the available delay in the filter. SO, when people hear differences (if they do), it would be more related to TIME delay instead of PHASE.

As soon as a person gets caught up into the audibility of phase, then things get confusing and controlled experiments start ending up being research projects. Since hearing is generally accepted to be SOMEWHAT phase insensitive, the difference in sound must be relative time delay. Our hearing is definitely time-delay sensitive, because we use that for localization of sound sources.

If you have short filters that might only have 100 samples of delay at 44.1kHz, that MAXIMUM amount of delay is going to be less audible than if there is a 1000 or 2000 sample delay. Those long delays, when the filter delay isn't constant vs. frequency, open up the possibility for large differences in wave propagation through the filter.

Here, I am not really arguing FOR the audibility, but as long as one keeps away from the 'phase' argument, and starts thinking about 'time' -- that is probably a better domain for an audibility.

*I just do EE, CompSci, DSP stuff -- I know very little about how people hear -- I am just talking about possibilites, not coming to any conclusion. I also suck at writing, so I apologize for my difficult-to-read prose...

John
 

maty

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...Now, with Win 10 highly optimized for multimedia the difference is more noticeable. Much of the improvement is due to the optimization of the access of the ALU to the internal records of the CPU, to the priority of the processes and to the management of the RAM.


Just now:

https://www.diyaudio.com/forums/music/6958-playing-listening-post6239677.html
It was sounding somewhat shrill. Turns out I had not run the last step of optimizing my Win10 for multimedia, which is manual.

What a difference right now!

diyaudio.com Lou Donaldson - The Time Is Right (1962), Vinyl x2, Analogue Productions 2010, 45rpm, US
 

maty

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Be aware that I just greatly improved the bass reproduction of the modified AVR I use. Along with the last touch-up that I just did to my little coaxial cables, I have a better sound in every way than yesterday when I played:

Melissa Stylianou - No Regrets (2014), CD, Anzic Records, US
https://www.diyaudio.com/forums/music/298646-album-listened-post6274496.html

And without spending a penny!
 

Julf

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Be aware that I just greatly improved the bass reproduction of the modified AVR I use. Along with the last touch-up that I just did to my little coaxial cables, I have a better sound in every way than yesterday.

If you say so...
 

maty

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Off topic

I am so satisfied that I just created an image copy of the windows partition with Acronis.
 

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