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Minimum Phase vs Linear Phase

ReaderZ

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Whilst I probably agree with what John is saying here, again it has to be put into context. How audible is an effect of moving the phase of say high frequency harmonics of a percussive sound? How much audible impact is there of changing the phase of a harmonic at 20 kHz that might be -70dB down?

Hmm my interest in filter and the reason I am reading this thread is my speaker doesn't have one of these ribbon tweeters that's designed to go to 45khz, spec shows it's only for 21khz. So I am worried about sending tons of ultra sonic signal that it's not designed to handle will likely will affect the sound it produce below 20khz.
 

March Audio

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Hmm my interest in filter and the reason I am reading this thread is my speaker doesn't have one of these ribbon tweeters that's designed to go to 45khz, spec shows it's only for 21khz. So I am worried about sending tons of ultra sonic signal that it's not designed to handle will likely will affect the sound it produce below 20khz.
Old thread so I cant remember the context.

Can you tell me what ultrasonic noise (and from where) you are concerned about?
 

MRC01

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... I am worried about sending tons of ultra sonic signal that it's not designed to handle will likely will affect the sound it produce below 20khz.
Could be. Ultrasonics are inaudible themselves but may trigger IM distortion in the amp, preamp or speakers downstream from the DAC. These are difference frequencies so they can be in the 1k range plus or minus where our hearing is most sensitive. For example I've seen some a dual tone 19+20 kHz test tones generate some nasty midrange distortion.
However, it all depends on the level of the ultrasonics. If they're sufficiently attenuated it won't be a problem.
 
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Hmm my interest in filter and the reason I am reading this thread is my speaker doesn't have one of these ribbon tweeters that's designed to go to 45khz, spec shows it's only for 21khz. So I am worried about sending tons of ultra sonic signal that it's not designed to handle will likely will affect the sound it produce below 20khz.
You're probably thinking about it in the opposite way than you should.
Tweeters with wider bandwidth might be bothered by ultrasonic information....not the other way around. If your tweeters head south at 21khz then high frequency information from weird filtering, or Class-D power amps, or other sources are not going to bother them.

Dave.
 

ReaderZ

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You're probably thinking about it in the opposite way than you should.
Tweeters with wider bandwidth might be bothered by ultrasonic information....not the other way around. If your tweeters head south at 21khz then high frequency information from weird filtering, or Class-D power amps, or other sources are not going to bother them.

Dave.

I see, make sense, I read somewhere someone blow his Adam tweeter from playing ultrasonic test tone.
 

ReaderZ

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Could be. Ultrasonics are inaudible themselves but may trigger IM distortion in the amp, preamp or speakers downstream from the DAC. These are difference frequencies so they can be in the 1k range plus or minus where our hearing is most sensitive. For example I've seen some a dual tone 19+20 kHz test tones generate some nasty midrange distortion.
However, it all depends on the level of the ultrasonics. If they're sufficiently attenuated it won't be a problem.


That's why I am looking at this thread and finding out why should I not just use the filter that goes down fastest/earliest for least amount of ultrasonic signal out of a DAC.
 

ReaderZ

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Old thread so I cant remember the context.

Can you tell me what ultrasonic noise (and from where) you are concerned about?


I was thinking I should just use the filter cut the most ultrasonic one. Then I find this linear, min, hybrid, and there is fast and slow etc, on other thread and some link to other site about how faster filter create more rings. Now I am reading this linear vs minimum where one create pre and after ripple and one have only after ripple.

Since slow = less ripple, why not a filter that start well before 20khz but has a gentler shape? Not like I can hear anything above 16Khz anyway. See filter 6 below, also it says hybrid, what does that mean?

SMSL M500 DAC and Headphone Amplifier Filter Response Audio Measurements.png
SMSL M500 DAC and Headphone Amplifier Filter Response Audio Measurements.png
SMSL M500 DAC and Headphone Amplifier Filter Response Audio Measurements.png
 

March Audio

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You're probably thinking about it in the opposite way than you should.
Tweeters with wider bandwidth might be bothered by ultrasonic information....not the other way around. If your tweeters head south at 21khz then high frequency information from weird filtering, or Class-D power amps, or other sources are not going to bother them.

Dave.
This.
 

March Audio

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I see, make sense, I read somewhere someone blow his Adam tweeter from playing ultrasonic test tone.
Yes there is no accounting for stupidity ;) This would have been at very high levels which will be an order of magnitude above any ultrasonic noise from dacs, amps etc.
 

March Audio

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I was thinking I should just use the filter cut the most ultrasonic one. Then I find this linear, min, hybrid, and there is fast and slow etc, on other thread and some link to other site about how faster filter create more rings. Now I am reading this linear vs minimum where one create pre and after ripple and one have only after ripple.

Since slow = less ripple, why not a filter that start well before 20khz but has a gentler shape? Not like I can hear anything above 16Khz anyway. See filter 6 below, also it says hybrid, what does that mean?

View attachment 60131View attachment 60131View attachment 60131
Ignore the pre-post ringing/ripple. Its a red herring. It occurs with signals close to nyquist, ie already above your hearing range. see below.

https://troll-audio.com/articles/filter-ringing/
https://www.audiosciencereview.com/...inimum-phase-vs-linear-phase.8762/post-220457
 

RichB

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I was thinking I should just use the filter cut the most ultrasonic one. Then I find this linear, min, hybrid, and there is fast and slow etc, on other thread and some link to other site about how faster filter create more rings. Now I am reading this linear vs minimum where one create pre and after ripple and one have only after ripple.

Since slow = less ripple, why not a filter that start well before 20khz but has a gentler shape? Not like I can hear anything above 16Khz anyway. See filter 6 below, also it says hybrid, what does that mean?

View attachment 60131View attachment 60131View attachment 60131

At 44.1 kHz, the Nyquist frequency is 22.05 kHz is the highest frequency from the source. The rest is dross.

- Rich
 

MRC01

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That's why I am looking at this thread and finding out why should I not just use the filter that goes down fastest/earliest for least amount of ultrasonic signal out of a DAC.
The filter's goal isn't to reduce HF, because most of it is part of the music. The key is to eliminate (or at least reduce) the portion of HF that is not part of the music -- that is noise. Theoretically, if the filter's stopband is Nyquist (22,050 for CD) then there won't be any HF noise.

Early / slow rolloff filters do have less passband amplitude ripple. But this is irrelevant because the amount of passband ripple with a well implemented fast rolloff filter is inaudible - hundredths of a dB. In that sense, a slow rolloff filter can be a cure that is worse than the disease. However, if the slow rolloff filter is applied at higher sampling rates, it can be perfectly flat up to 20 kHz which makes the slow rolloff harmless.

Take at look at the above filter graphs. 3 of them (Magenta, Blue and Brown) are fully attenuated by Nyquist. None of these will have any HF noise. Magenta is slow rolloff which is unnecessary. Blue or Brown is the one to pick - one probably is linear and the other minimum phase. Linear is the most "correct" yet some people prefer the sound of minimum phase. Most people can't hear any difference.

Now look at the two filters (red and teal) that don't fully attenuate until 24,100 Hz. These filters use a common engineering trick. They take advantage of the fact that each frequency's alias reflects around Nyquist to ensure that any and all HF noise it leaks through will be above 20,000 Hz. In other words, Nyquist (22,050) is the exact center of the filter's transition band (20,000 to 24,100). So there's no audible noise, and this gives these filters a transition band that is twice as wide, so they are cleaner and flatter in the passband. However, suppose this filter leaks energy at 23,000 Hz. The alias is 22,050 - (23,000 - 22,050) = 21,100 Hz. Both tones are inaudible. But their difference tone: 23,000 - 21,100 = 1,900 Hz is easily audible. So as these 2 supersonic tones pass through the analog preamp, power amp, and speaker/headphone, intermodulation distortion will create a 1,900 Hz tone. That's the bad news. The good news is that the filter attenuates these frequencies around 20 to 30 dB, and IM distortion is typically at least another 60 dB lower, so in most cases the 1,900 Hz distortion tone will be inaudible. And low enough in level that it won't hurt tweeters.

Overall, given the above filter choices, I'd use the red labeled "fast linear". It should be the most transparent in the passband and ensures that any HF noise is very low in level and above 20 kHz. But if you want to be super conservative about not allowing any HF noise at all, use the Blue (apodizing 5) or Brown (brickwall 7).
 

DonH56

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John Dyson

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One item of note about 'pre ringing'/'post ringing' in the case of othwrise flat FIR type filters... They don't 'ring', but one mostly sses the differences of the available frquency compnents. The simplest case is a linear phase with fast rolloff. If you supply a square wave, the 'lore' says that it will 'ring'. That isn't quite true, as the 'ringing' is actually 'Gibbs effect', which happens because of MISSING frequency components and not really stored energy like a high 'Q' filter might. 'Gibbs' isn't really ringing -- think of it as the effect of missing high frequency components that would have otherwise cancelled out the 'wobbles'. It is a residue of sorts.
When flat filters not-linear-phase have the 'wobbles' move from one place to another in the resulting waveform, that 'move' is actually the result of the timing at different frequencies not being the same.
Any audible difference between a non-ringing filter *MIGHT* be because of different propagation delays of different frequencies. Lower frequencies will usually propagate with a different delay from the higher frequencies for filters without constant delay like a linear phase filter. I am not 100% convinced that there is an audible difference between short constant delay vs. non constant delay filters, but can be convinced. Non constant delay flters with more taps might be easier to intellectually convince. (Longer availabel delay in the filter -- more space for timing delay differences.)

John
 

hyperknot

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I have a quick question, without reading through the whole thread (sorry if it was answered alread): There are many devices now which have problems with their built-in filters, like balanced DACs from SMSL and Soncoz recently.

Now my question is: if I'm only planning to use software based upsampling, that is, my DACs would never receive anything less than 96 kHz, do I need to worry about the built-in filters' problems? I'm not planning to do PCM -> DSD conversion, just simple PCM upsampling from 44 -> 96 kHz.

I mean if I feed 96 kHz, will it push the built-in filters' problems to over 48 kHz? Or should I upsample to 192 kHz to push that into the 96+ kHz range?
 

hyperknot

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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?
 

bennetng

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I have a quick question, without reading through the whole thread (sorry if it was answered alread): There are many devices now which have problems with their built-in filters, like balanced DACs from SMSL and Soncoz recently.

Now my question is: if I'm only planning to use software based upsampling, that is, my DACs would never receive anything less than 96 kHz, do I need to worry about the built-in filters' problems? I'm not planning to do PCM -> DSD conversion, just simple PCM upsampling from 44 -> 96 kHz.

I mean if I feed 96 kHz, will it push the built-in filters' problems to over 48 kHz? Or should I upsample to 192 kHz to push that into the 96+ kHz range?
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.
 
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