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Minimum Phase vs Linear Phase

maty

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[PDF] https://www.rme-audio.de/download/adi2profs_e.pdf

RME-ADI-2-DAC-FS-filters.png


RME-ADI-2-DAC-FS-impulse-1.png


RME-ADI-2-DAC-FS-impulse-nos.png


My traduction:

* If the RME DAC works in a studio recording -> Short Delay Sharp

* If the RME DAC works to enjoy music -> Short Delay Slow

SD Slow causes a small drop in the higher frequency range, but has a less aggressive (less steep) filter

With a good/very good 64-bit PEQ -> flat frequency response. But someone likes more this response at the listening point.

index.php



PS: I guess RME engineers do not know anything either, I say.
 
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daftcombo

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[PDF] https://www.rme-audio.de/download/adi2profs_e.pdf

View attachment 33249

View attachment 33250

View attachment 33251

My traduction:

* If the RME DAC works in a studio recording -> Short Delay Sharp

* If the RME DAC works to enjoy music -> Short Delay Slow



With a good/very good 64-bit PEQ -> flat frequency response. But someone likes more this response at the listening point.

index.php



PS: I guess RME engineers do not know anything either, I say.

Nice to see those documents again but I wonder what's going on with your "traduction".
 

maty

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Addenda: the physical world, at least in audio, has a minimum phase response, including speaker drivers. Logically, the reconstruction (digital -> analog) in the DACs must be of a minimum phase (fast/sharp or slow).

Digital filters at the studio, usually are linear phase.

Then, by default, the logical thing is that it is the minimum phase. Then those who love each other are added, because today music is created and manipulated looking for spectacularity and not fidelity and that is what the vast majority seeks, such is the sad reality today.
 
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maty

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Off topic

I usually use the word emotion but others also handle similar concepts or feelings. Today I have read a comment that made me think about writing this off topic:

AVA SET 120 amp pleases a retired audio reviewer
https://www.audiocircle.com/index.php?PHPSESSID=j2easgo8ate5cag255b2bfbhf6&topic=165527.msg1759170
That last statement says it all... "When I turn on my system, I don't want to be blown away, I want to be drawn in, to be engaged, not entertained." Sounds like a tag line in the making...

Some of us do not look for spectacularity but rather get excited more and more frequently when listening to good music recordings, and not simply stay with the mouth open due the great sound. Some listen to music and others devices or musical products, I say.

- End off topic -
 

MRC01

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So I've carried some prior knowledge into this and also learned a few new things. To summarize: there are 2 variables with the AA filter: transition band width (narrow/fast or wide/slow), and linear vs. minimum phase. These are independent of each other, giving 4 basic categories (each with with variations since you might add different windowing functions).

Some here seem to assume the "sharp" filter is always closest to the ideal sinc(t) performance, and that "slow" filters are always minimum phase with more distortion. That's not necessarily the case! Sometimes the default "sharp" filter is sharper than it needs to be. Sometimes the "slow" filter is not minimum phase.

A slow filter with linear phase should be ideal and get closest to sinc(t) performance. By "slow" I mean as slow as possible, but no slower. By that I mean, make the transition band as wide as possible without attenuating in the audible bandwidth. For example, always start the filter transition band at or near 20 k no matter high Nyquist is. There's no reason to have flat response into supersonic, and you can use the extra range to give the filter a more gradual slope having more ideal response.

My prior test was with tones generated at 88.2 and 96 kHz. Since we have a rainy weekend coming up, I'm going to do another test with tones generated a 44.1 kHz and post the results / comparison here.
 

BDWoody

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Off topic

I usually use the word emotion but others also handle similar concepts or feelings. Today I have read a comment that made me think about writing this off topic:

AVA SET 120 amp pleases a retired audio reviewer
https://www.audiocircle.com/index.php?PHPSESSID=j2easgo8ate5cag255b2bfbhf6&topic=165527.msg1759170


Some of us do not look for spectacularity but rather get excited more and more frequently when listening to good music recordings, and not simply stay with the mouth open due the great sound. Some listen to music and others devices or musical products, I say.

- End off topic -

I listen to my music through the required devices. As long as the devices are competent, they won't get in the way of the emotion communicated through the music. I start tweaking, I have no idea what I'm going to end up with.

Some like to feel that they can improve on everything, then not so remarkably 'hear' all the differences they have been expecting all along...then get upset when others ask for more than just insistence...

So many claims...so little evidence.
 

maty

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Some here seem to assume the "sharp" filter is always closest to the ideal sinc(t) performance, and that "slow" filters are always minimum phase with more distortion. That's not necessarily the case! Sometimes the default "sharp" filter is sharper than it needs to be. Sometimes the "slow" filter is not minimum phase...

Both, Short delay RME filters are IIR and minimum phase. Default: sharp/fast, which is the most appropriate to listen to deficiencies in mastering, hence the need for a flat response at the point of listening. But you know -> usually hearing fatigue. One thing is work and another enjoy.
 

MRC01

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As I understand it, "Short delay" often means minimum phase, because it doesn't require looking ahead, which linear phase does. That looking ahead means pre-buffering samples, which causes latency.
 

maty

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At fs=176.4, Nyquist is 88.2.
The sharp filter response passband goes to 0.4fs, so it's flat to 70.56 and transition band is narrow (88.2 - 70.56 = 17.64).
The slow filter response passband goes to .104fs, so it's flat to 18.35 and transition band is wide (88.2 - 18.35 = 69.85).
The slow filter has a transition band that is about 4x as wide. Its gentler slope creates less passband ripple.

It the delay is less than... 1 ms (I do not remember just now what the minimum value is) you can not hear it.
 

maty

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https://www.google.com/search?q=audio+delay+minimum+value+hearing

-> http://www.hearingreview.com/2010/01/acceptable-processing-delay-in-digital-hearing-aids/

One should keep in mind the researchers’ caveat that users may actually tolerate longer delays from real hearing instruments (ie, above the 5 to 6 ms values their study suggested). However, the extent that a delay is “noticeable” does differ somewhat among individual patients, and is related to degree of hearing loss, cognitive status, and other factors. Therefore, it is also possible that, for isolated cases, an even shorter delay than shown in Figure 2 for the Siemens Pure might be required to achieve optimum sound quality. In such instances, we recommend disabling some of the adaptive signal processing features, which add to hearing aid delay. Switching to a device with fewer channels, and therefore fewer filters, would also be a reasonable solution...

But now the music usually only has synthetic instrumentation.
 

scott wurcer

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It the delay is less than... 1 ms (I do not remember just now what the minimum value is) you can not hear it.

Wrong use of "delay". You put the CD in and say the music starts 3 sec. later the processing delay is simply added to that. It's not like video and audio out of sync.
 

mansr

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Best to use music and @pkane differential software.
Sweeps and impulse/squarewaves are nice test signals but that's all they are.
The impulse response fully characterises a linear system. Tones and sweeps are useful to determine non-linear distortions. Music is a terrible test signal for any purpose. If you need something non-repeating, use random noise.
 

MRC01

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Here are the 44k test results for difference between slow & sharp filters. My test method is not ideal but it shows some differences. I generate the files in REW, burn to disc, play through my DAC (once with sharp filter, once with slow filter), record its balanced analog outputs to my Tascam SS-R1 recorder. Then analyze the WAV files from the recorder. The recorder certainly has its own FR, distortion, etc. so the absolute values don't mean much; what matter is the differences between the files.

Below: dark blue is "sharp", light blue/cyan is "slow". When I couldn't put both on the same screen, I always list sharp first, slow second.

First, frequency response. The slow filter rolls off earlier but the difference should not be audible (at least not to me).
44-fr1.png


Let's zoom in on that top octave.
44-fr2.png


These charts show no difference in distortion (sharp, then slow)
44-dist-sharp.png


44-dist-slow.png


Group delay. As expected, a difference in high frequencies but it is so small as to be inaudible.
44-gd1-sharp.png


44-gd1-slow.png


Let's zoom in on that high frequency group delay difference. The difference turns out to be above 20 kHz, so nothing to hear, here.
44-gd2-sharp.png


44-gd2-slow.png


Now for the spec where I expected to see a difference: impulse response. The slow filter rises faster and decays slower, but not by much. I expected to see more difference here.
44-impulse1-sharp.png


44-impulse1-slow.png


Here are the envelopes on these responses. I don't see much difference here. The slow has less ripple, but if you count bumps per division the ripple is at 22 kHz.
44-impulse2-sharp.png


44-impulse2-slow.png


Here's the step response. Sharp has more pre-ripple, slow has more post-ripple. But the overall shape is virtually identical.
44-step1-sharp.png


44-step1-slow.png


Finally, here are CSD plots -- no significant differences I can see.
44-csd-sharp.png

44-csd-slow.png


This doesn't show much in the way of differences. But even that might be helpful to people interested in this subject.
 

MRC01

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Here's another one, even more interesting. I figured I'd see a difference in a square wave and I was right:
First, sharp:
1568429514768.png

Now, slow:
1568429564790.png


The sharp filter ripples less, but it ripples almost equally before and after the steps. The slow filter ripples more after a shift, overshooting and taking longer to stabilize. But it has no ripple before a shift.
 

solderdude

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The 'fun' part is that you are showing signals that don't exist in any music files.
All 44.1 recordings should have passed through a steep brickwall filter on the encoder side (when down-sampled properly from the recording format) so frequencies that are reaching Nyquist should not be there. Square-waves that were recorded through the analog input or are down-sampled won't have such steep rising and falling edges.
Besides, the upper treble 'energy level' in recordings (almost) never reaches near FS. So stuff that 'mirrors' back due to slow filtering also will be lower in amplitude (unless one is daft enough to use filterless at 44.1/48)

That's why I suggested to use (recorded) music and nulling software if you want to find out where the audible differences are.
One should realize though that phase shifts, in nulling, are converted to amplitude differences as well.
So some of the heard differences may come from inaudible phase shifts but in reality may be mostly inaudible IRL.
We all have seen the impulse, square-wave, white noise and IMD plots for various filters and the idiotic conclusions some folks draw based on plots they don't understand.

Otherwise, if you really want to know try controlled and properly conducted blind tests. Tedious, exhausting and requires someone else to help you and a statistic relevant number of 'tests' to be conclusive.
Your attempts may show other results from those of someone else as well.

For me it is 'clear'... steep linear phase is the only way.
The fact that some folks 'prefer' another or no filter and claim superior sound has little to do with how it is done properly.
 

maty

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maty

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https://www.google.com/search?q=mytek+dac+reconstruction+filter

-> [PDF] https://mytekdigital.com/download_library/reviews/Stereophile_Mytek_Brooklyn_Review_November2016.pdf

Mytek-Brooklyn-dac-stereophile-filters.png


RME until now their products were focused on professional audio. MP sharp/fast is the default.

Mytek is more Hi-Fi audio. And you know, MQA. MP slow is the default.

Benchmark, I think it works after a own oversampling. LP.


Unfortunately, for decades Hi-Fi is almost always Hi-Sp, with Sp from Spectacular and not from Spain ;)
 
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RichB

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Anyone with an Oppo UPD-205 can do a SBT test. The media control app can be used to switch the filter between say, Linear Phase Fast and Minimum Phase Fast. They can be in the other room as well.

- Rich
 
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