On the subject of "ringing"

[buonassi]

IMO preringing matters greatly on headphones at least.  Perhaps the effects of "ramp up" or preringing are not as apparent when played on speakers in a free field environment.  But when the drivers are an inch away from your ear, the timing becomes very important.  I have a lot of respect for @Archimago. But I read on one of his articles (on his blog) that preringing is only apparent at high frequencies.  My ears disagree.

 

The preringing introduced by digital filter reconstruction seems to affect all frequencies.  Or perhaps it only changes the time alignment of the highs in relation to lows.  Which is happening I can't say. What I can say is that I prefer minimum phase filtering to linear phase.  For me, transient alignment is paramount to Rock/Metal/Prog music I enjoy so much.

 

It's like any other audio phenomenon:  when you hear it, you can't unhear it.  If you know what to listen for, you can pick out preringing on headphones pretty easily provided you have a decently resolving system.  For me, the biggest tell of preringing is that instruments separation is almost "too good".  While the stage becomes very expansive and space between the instruments is increased, you give up some precision on the attack.  There is definitely a tradeoff here.  

 

The next biggest tell is in an instrument that covers both high and low frequency in the same transient:  The bass kick drum.  A kick drum should have a click or snap of high mid frequency (when the batter head strikes the drum head) followed by a decay of bass and sub bass as the note decays.  If the transients are aligned using a linear phase filter, what you hear is a slight "chuffing" of the sub bass coming in just before the click of the batter head.  Some call this "ramp up".

 

I believe we've been listening to music on CDs for a long time and our brains have become accustomed to this ramp up sound.  Plus, it's not overly apparent unless you are listening critically.  Once you hear it though, you won't unhear it.  The overall effect of kick drum timing of high vs low frequency results in a blurry presentation, where that drum gets buried in the mix and the attack is subdued.  The same can be observes for all instruments to some degree.  I only use the bass kick as the most evident example. 

 

So take your pick:  enhanced instrument separation and open staging (linear phase) or perfectly timed transient attack (minimum phase).  You can't have both at 44.1 and a flat bandpass of 20-22k.  Now, if you're willing to concede some of the high frequency response, and design a filter that slowly rolls off, you can have the best of both worlds (but it'll sound dull at 44.1 pcm)

 

What I've just burped out is the case for high res PCM or DSD.  You can eliminate pre and post ringing by having a very slow rolloff filter ABOVE the audible range, leaving a perfectly reconstructed, time aligned sound between 20-20k.  

 

Long video, but it helps describe and give an example of pre-ringing in a very exaggerated form.  Note that the author chose to use the bass kick as the instrument to demo these preringing effects:

 

[buonassi]

4 hours ago, John_Atkinson said:

 

Incidentally, I have recently been examining the time-behavior of A/D converter antialiasing filters and have found just one which captures a band-limited impulse without any ringing before or after: the Listen filter on Ayre's QA-9 converter, designed by Charley Hansen and Ariel Brown.

 

John Atkinson

Editor, Stereophile

 

Thanks,  I should've noted that the ringing can also be a result of the encoding process as well as the decoding process.  

But isn't this ADC you mention achieving less ringing at the expense of high frequency loss?  I thought the ayer "listen" filters (at least in their DACs) were a super slow rolloff starting at about 15K?  It's the same for any DAC that uses the AKM4490 chip (note pic below).  

 

Presumably, when using both slow rolloff at encoding, then again at decoding, you're attenuating the top octave pretty aggressively....

 

 

[buonassi]

@John_Atkinson, Fiio X5 3rd gen, uses a filter called "super slow rolloff" that produces almost no ringing.  It must be proprietary to them because it's not listed on the AKM data sheet.  But I can clearly hear the high frequencies being attenuated when switching between this setting an a more traditional filter of the 4490.  Admittedly, this device's amp section really isn't up to snuff for the most critical of listening, but still the result of this special filter is very dry sound - eerily precise. But again, at a cost....  HF rolloff.

http://fiio.me/forum.php?mod=viewthread&tid=42106&page=1&extra=#pid117558

 

On a side note, I've somewhat given up on trying to eliminate ringing.  A minimum phase filter that dumps all the ripple energy after the transient is OK by me provided it's not excessive.  I've played around with iZotope SRC at length to find a good compromise between high frequency loss and post ringing - still avoiding aliasing at -96dbfs.  And of course, I never stray from the minimum phase region of the ringing parameter.  

 

Really it's the "pre" ringing that bothers me.  Post ringing has the effect of added harmonics on cymbals, giving them some more zing and sparkle, which can be overwhelming if too much post ringing is present.  A moderate amount really doesn't bother me.

 

 

 

[buonassi]

8 minutes ago, John_Atkinson said:

The response at 20kHz is down by just a fraction of a dB at all 3 sample rates, though the rolloff is slow, which will potentially allow some aliasing with some music.

 

And therein lies the tradeoff: aliasing.  The "filtering" power of this seems very weak for red book material if we're talking about DACs and reconstruction filters for playback.  But this is ADC only we're discussing for the QA9, right?  Not SRC/downsampling from a digital master to redbook - where aliasing would matter most.  I'm too ignorant on ADC to understand why the slopes and bandpasses of the filters are essentially identical for all sampling rates.

 

Still, good article - I especially agree with your sentence on page 1 - 

 

"Given my extensive experience of both domestic and professional A/D converters, which has convinced me that the most critical process in digital recording is the initial analog/digital conversion—nothing downstream can put right whatever was done wrong in that conversion"
 

[buonassi]

16 hours ago, Fokus said:

 

That is a demonstration of equalisation with the filter transitions smack in the audible band. It has been known for three decades or so that under these conditions, given sufficiently steep transitions, linear phase pre-ringing can be audible.

 

But if you move the transition frequency up and up the audible effect vanishes.

 

Not sure it vanishes, but it's subdued for sure.  In terms of digital reconstruction filters on DACs, there isn't the same level of OVERT preringing, of course not.  It is very subtle, but it's there.  It can be measured, but that's really not the point.  If I'm inferring correctly, I think your assertion is that the human auditory system can't discern preringing of such low levels and therefore any preringing in digital filter design is of no consequence? 

[buonassi]

7 hours ago, John_Atkinson said:

 

As its frequency is at Nyquist, which is above almost everyone's HF cutoff frequency, you would think not. But when I discussed this with Karlheinz Brandenburg at an AES convention several years ago, he said that basically even if you can't hear the "ringing" as a tone, your brain could well be aware that something has happened when it starts and marks it as an acoustic "event." Then, when the peak is subsequently reached, that is marked as a spurious second "event," leading to confusion.

 

 

Does preringing only affect HFs near Nyquist?  The entire bandpass is affected I thought?  Preringing has less to do with the frequency domain, rather the time domain as I understand it, and all frequencies in bandpass are affected in the time domain regardless of Nyquist limit.  

 

I agree with your second sentence, which time domain theory would support.  Preringing doesn't exist naturally.  You can't "ramp up" to a transient.  That would necessitate breaking the laws of physics to somehow get a portion of the decay of a transient to your ears before the original transient takes place. When a linear phase transient-aligned filter is used, some information is pushed ahead of time during reconstruction, modeling precisely what I just described.  Unnatural. And your brain can hear it IMO.  

 

So, is unnatural worse?  No, it's just different and could be considered quite enjoyable by some people.  I like it with acoustic/orchestral music actually!  But for metal, with super fast drum fills, not a chance.  

[buonassi]

11 minutes ago, Spacehound said:

It doesn't happen at all if Nyquist/Shannon is obeyed.

 

are you sure you're not confusing aliasing/imaging with "ringing"?  Not trying to shut down the discussion, or insult you by any means.  It's me that has a lot to learn yet, especially about Nyquist and the "folding" that's involved with MQA.

[buonassi]

25 minutes ago, Spacehound said:

I don't think I am but I wouldn't stake my life on it. It's only  audio, not Klingon babies

Pre-ringing  breaks cause and effect so it can only be an artefact,  or what you see if you use an invalid measurement method. Which I suspect they are doing deliberately.

The effects of preringing can be done away with just by using a higher sampling rate with a super slow rolloff filter applied after 20khz.  So a benefit of MQA as I see it would be a smaller file size.  But do we really need that with today's storage and transmission technology?

 

Man do I wish SACD or even 96khz/16 bit PCM would've become the standard and ousted redbook.  We wouldn't be having this debate today.

 

edit:  The effects of ringing both pre and post can be done away with.

[buonassi]

just got caught up.  in reading the last couple pages, I keep seeing attempts to prove ringing doesn't exist or shouldn't matter if it does.  I'm not even going to try and pretend I can follow some of the posts as I'm not a physicist or electrical engineer.   

 

if ringing doesn't exist, or doesn't matter, then why are there different filter designs and why do they sound different to me?  Are the opponents of ringing going to assert that I'm imagining the differences?  That "expectation bias" is to blame?  

 

I'm not going to state that one filter is better than another.  But am I the only one who can admit to hearing a difference among them?  Have all the hours I've scrutinized iZotope upsampling parameters, observing the rolloff steepness in an RTA, and noted sonic differences all been a hard lesson in realizing I'm "drinking the coolaid"?

[buonassi]

5 minutes ago, Hifi Bob said:

Ringing does exist, as I showed in the spectrograms above, but it seems only with very steep filters.

 

If you can hear it, then it’s probably due to a non-linearity in your playback chain—possibly a damaged tweeter.

single dynamic driver per channel headphones - focal elear.  Same on my HD600s and even on my phase aligned balanced armature in ears (JH/AK Angie).  

 

A properly setup speaker system (sans sub) should uncover ringing given that frequencies above 100hz travel roughly at the same speed.  Yet I read somewhere that it's not as evident on a free field speaker system given reflections, room effects, standing waves at low frequencies, etc.  But that was purely conjecture.  

 

[buonassi]

are "ringing" and "time smearing" the same concepts?  

[buonassi]

being my 101th? no, that's not right, 101st post, I have taken the liberty of graduating myself from "freshman member" to to this..... promulgating which camp i'm in.  Damn do I feel assertive tonight.  

[buonassi]

 

14 hours ago, Spacehound said:

 

In practice? If you cant  hear any difference buy the lower cost box.

That's why most  Hifi manufacturers HATE blind tests and will come up with near endless utter nonsense to (fail to) convince sane people they don't work. It is why most US manufacturers (particulary)  have totally abandoned  the  realms of high fidelity and gone for 'impressive' instead.

 

well said.  I totally agree with your comment, and you know what?..... I can't help but ask myself this when i eval gear at home:

 

"If I didn't have them side by side, could I tell you which was which?  am I really missing any musical enjoyment by foregoing the more expensive unit"?

 

my answers are almost always "no and no" as I climb higher up the chain...... but.....

 

My point is..... I buy two of something when I make a purchase (ie headamp).  I later I sell the one I like less after courting them for a nice long time.   I find that differences are revealed as you became more "tuned in" to each unit's presentation and subtleties - and this takes time.  For me at least, once I hear something (like or dislike), my brain becomes trained and the affinity or aversion to what I'm hearing becomes much stronger.  Yes, "ringing" I'm looking at you.  I'm clearly forming a preference (subjective) and it's one that I can discern (again subjective).  What was first answered as "no and no" becomes "yes and yes" the longer of time you spend with those units.  

 

This hobby, or in my case, audiophile curse, is all about a few percent of subjective SQ improvement with each change.   And let's agree on this at least:  2% here and 3% there add up to a final chain that brings joy and satisfaction....for a little while at least. 

 

[buonassi]

On 2/10/2018 at 7:40 PM, buonassi said:

The next biggest tell is in an instrument that covers both high and low frequency in the same transient:  The bass kick drum.  A kick drum should have a click or snap of high mid frequency (when the batter head strikes the drum head) followed by a decay of bass and sub bass as the note decays.  If the transients are aligned using a linear phase filter, what you hear is a slight "chuffing" of the sub bass coming in just before the click of the batter head.  Some call this "ramp up".

 

I think my understanding has matured somewhat here.  Looking for others to validate this if possible.  I wrote the above a little over a month ago and have been racking my brain to understand the MP vs LP differences, both in critical listening and in reading.

 

What I think at this point is that I prefer minimum phase because I actually like the phase distortion.  It is the phase distortion that is separating the attack "smack" of the drum verses the resonance of the drum.  They are no longer perfectly aligned, and therefore the smack is easier to hear.

 

But I got the order wrong.  I'm pretty sure after reading that the HFs actually arrive shortly after the LFs when a MP filter is used, so what I'm likely hearing (and liking) is the separation of the drum attack vs its resonance, regardless of which comes first.  Because these transients happen so fast, my brain can't really tell if the smack is coming before or after the kick drum's bass resonance.  All my brain hears is better separation, and therefore better perceived attack, even if the smack is happening shortly after the bass bloom.

 

So much for my "chuffing" comment.  Gotta eat some crow on that.....

[buonassi]

On 3/22/2018 at 6:46 PM, semente said:

 

This is quite interesting because to me it sounds like a (special) "effect", and I don't like it because it sounds less real or natural with the kind of (acoustic) music I listen to. But I found the 'ASYM' filter an "interesting" alternative to 'LP'.

 

@Miska does recommend 'LP' for acoustic music and 'MP' for pop/rock for a reason.

 

The ASYM is for asymmetrical , meaning "intermediate" phase right?  I've had pretty good results with this type of a filter as well.

 

I started going back and reading some of @Miska's older posts about his filters in hq player.  What I came across confused me even more.  Originally, I believed it to be the case that low frequencies were delayed realtive to the highs in a minimum phase filter-because that's what I was hearing.  Then, I was pointed in other posts that I may have it backward, it's the highs that are delayed relative to the lows.  Now, after reading this I'm back to believing the lows are delayed, which sounds more natural to me (bass kick drum example).

 

"Linear phase filter" is a filter where all frequencies pass with same time delay. "Minimum phase filter" is a filter where all frequencies pass through as fast as possible, higher frequencies faster than lower ones." https://www.computeraudiophile.com/forums/topic/13071-hqplayer-resampling-filter-setup-guide-for-ordinary-person/?do=findComment&comment=175928

 

 

Maybe an expert on the subject can just put this to bed once and for all.  From this statement I'm inclined to believe the lows are being delayed and arrive at the eardrum shortly after the highs do.  So which is it?  Highs or Lows delayed when using minimum phase?

[buonassi]

here's a fun one.  i've already invested an hour of my life I'll never get back here, trying to figure out the question to my 2 month old question:

http://www.sengpielaudio.com/calculator-timedelayphase.htm