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Speaker time alignment, does it matter?

Inner Space

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This is an interesting question. It goes back to the ( quite faulty ) foundations of stereo as a concept. It can be shown, that the human hearing reacts strongly to inter-aural time differences. That is the difference in time, at which some sound arrives at the left and right ear. The nearly same effect can be achieved by exposing the ears, again left and right to sound of different volume. These differences, time and volume, are perceived as an estimate on the direction, from which the sound comes in. Some physical phenomenon is used by the senses as to construct a picture of the surrounding.

This trained ability of the hearing is used with stereo, but very imperfectly. The often told theory behind it is hilarious nonsense. It assumes a person who is hearing straight forward and never moves its head. The audiophile in this delightfull stereo situation cannot easily be distinguished from a manequin. Actually, some folks tries to follow the rules :facepalm:

O/k, what stereo does is to induce and exploit those mentioned inter-aural differences in the studio as to mimic directional "information" by treating the "signal". But because the theory never holds, the sound engineer / mixer will better test listen to the recording. Only once she is satisfied with her settings, after many correctiions for good, it is published.

Stereo never ever reproduces the original sound field!

Back to Your question. The quite artifical situation of stereo listening is shown to not react to phase lag, excessive group delay, time misalignment, if they are identical in both speakers used. That is a matter of fact. It holds within some uncertain bounds, sure. But the limits are never hit with regular quality designs. Take it for granted, that You will not experience any drawbacks from phase eq. egd eq. tma etc whatever they name it. As long as it is the same for both speakers. It seems the hearing doesn't use the related physical phenomenon for creating a picture of the outside world.

Why is it all so complicated? Because stereo addictives from audiophilia often cite scientific investigations, which are not: not focussed on or even related to stereo. Group delay of so and so is perceptable blah blah and so loudspeaker have to ... (put in some hilariously excessive requirement). Such conclusions are plain wrong. They lack "the middle piece" in their derivations. They secretly assume that stereo has to reproduce the "signal", which is the original sound field. They say, because it is not the original "signal" the "information" is lost. Plain wrong, stereo re-invents the "information" on direction by introducing artificial add-ons to some excerpt of the "signal".

And that can be done and is only done as a difference between the two channels of the program material. As long as the intrinsic phase/volume ondulations of each speaker of a re-producing stereo pair are kept the same, these are irrelevant.

I think you're being a little harsh. Remember, "stereo" doesn't mean "two" or "pair" or anything numerical or quantitative. It's from the Greek stereos, which means "solid" or three-dimensional. The idea was to produce a three-dimensional sonic reproduction of a three-dimensional sonic event. The first experiments were 139 years ago, in 1881. They used two channels, but only because it was a headset-based system. (These trials were public, and attracted what was probably audio's first-ever subjective review, in Scientific American, no less: " ... the sound takes a special character of relief and localization ... to produce this remarkable illusion ... "

Later, Bell Labs experimented with up to 80 channels, before eventually settling on 3, along with most everyone else. Blumlein's insight was to start at the other end of the problem. Delivery was the issue. Two channels was the practical limit. So two channels had to work. And they did. The 90-degree crossed-pair microphones produced hundreds of examples of virtually perfect three-dimensional reproduction. To say, as you do, "Stereo never ever reproduces the original sound field!" is seriously misinformed. Two channels can do it, and I'm pretty sure 80 can, too.
 

tuga

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To say, as you do, "Stereo never ever reproduces the original sound field!" is seriously misinformed. Two channels can do it, and I'm pretty sure 80 can, too.

With speakers it doesn't, because all sound - source/direct and room/reflected - is coming from the two speakers, not from the original location. Clapping in a live concert recording is a good example.

x4WzYt1.gif




You'd need something like this to roughly reproduce the soundfield and a spherical mic with as many channels:

q6u1uiY.jpg
 

GelbeMusik

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Okay, to be truthful I wasn't able to follow all that, even though I read it four or five times.

My understanding is that what makes the L-R crossover slopes special is that the two drivers remain phase-coherent ...
Why would phase coherency of the two wavefronts induce ...?

I read the rest of it, several times. But I've already written more than I meant to and more than I should have.

I'm accused to be harsh. I'm not. I'm honest. There is no use in discussing things like these with no more of a background. I was referring to Your or any others college script. If You or somebody else would be able to explain what this is all about without referring to loudspeakers, that would motivate me to explain ( hint: what exactly is "now", "time" in this context? The answer may come out lengthier than one might whish for. Then You've got it right, right? )

Just rephrase the problem. Then we'll see.


Yes. But again, he refuses to go into any further detail. Why? Not because it is a secret, or he has something to hide ( conspiration? ). It is because advertising bombards people with technical terms witch they won't understand--on purpose. As to confuse people, while making them feel competent to actually decide to: buy their product. To really explain needs a few college level lectures in basic mathematics and physics. It is not possible to learn about such things in the context of loudspeakers alone. Sorry!
 

Frank Dernie

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It isn't exactly new, there have been time aligned speakers available since the 1970s so if it were a crucial aspect for SQ all speakers would be time aligned by now.
Simply miked material has always had impressive imaging IME, so a Blumlein crossed pair, Decca tree and ORTF stereo microphone system gave super solid stereo image and may, perhaps, benefit from time alignment, I have this impression from a demo of the effect of DSP speaker correction.
However most recordings nowadays are mixdowns of multi channel recordings so no phase coherency anyway so any stereo imaging is synthetic. The gain of the DSP correction demo was much, much smaller on modern multi microphone recordings.
 

KSTR

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What inter-driver time-alignment really means:
  • Send narrow-band tonal bursts (wavelets) through the speaker, record the individual driver outputs (other drivers muted, and with strict timing reference, of course). Tonal burst are sines windowed with, for example, a raised-cosine function, see "CEA-2010 bursts"). Use base frequencies at the XO freqs and nearby
  • Overlay the responses from the individual drivers, there must neither be a difference in the position nor in the waveform shape itself. The outputs shall be the same, apart from the level differences. Often, the position coincidence criterion can't be met, then we can widen the scope to: the position offset shall be strictly proportional to 1/f, that is, a constant phase offset is OK. Of course, only small offsets are useful, in the 0° to +-90° range, beyond that acustic summing is actually subtractive and lobing issues will be extreme.
Then, and only then, both phase and group-delay do track and the acoustic summing is perfect.
In other words, you can judge the timing aspect only once the wavelet output is identical in shape first.
 
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KaiserSoze

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As best as I could tell, what that Mr. Brunhaver is saying is that even if physical alignment of drivers is used to achieve phase coherency in the overlap region of two drivers that interface, this won't eliminate the overall rotation in phase that occurs in a speaker from low frequency to high frequency. So what? Does he mean to imply that since the overall rotation in phase cannot be avoided through physical alignment of the drivers, that there isn't any benefit to making the two drivers phase coherent in the overlap? The impression I'm left with is that he doesn't understand the reason that with most speakers the tweeter is vertically aligned with the rear of the woofer cone. This is done for reasons that have to do with the proper summation of the two individual wavefronts within the overlap frequencies, such that the main lobe, for any frequency within the overlap, will extend along a horizontal plane, rather than tilt up or down. If the lobe tilts up or down, then the tonality of the speaker will be different at different distances from the speaker. In order to avoid tilting of the lobe, it is necessary for the two individual wavefronts to be phase-coherent, not merely at the exact crossover frequency, but throughout the overlap region. Vertical alignment of the two drivers is one aspect of achieving this goal. It might not be essential to do this in order to achieve this goal, but when it is done, it is done toward achieving this goal, and it makes perfect sense, irrespective of the fact that there will still be an overall phase rotation from low to high frequency.
 

KaiserSoze

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I'm accused to be harsh. I'm not. I'm honest. There is no use in discussing things like these with no more of a background. I was referring to Your or any others college script. If You or somebody else would be able to explain what this is all about without referring to loudspeakers, that would motivate me to explain ( hint: what exactly is "now", "time" in this context? The answer may come out lengthier than one might whish for. Then You've got it right, right? )

Just rephrase the problem. Then we'll see.

Yes. But again, he refuses to go into any further detail. Why? Not because it is a secret, or he has something to hide ( conspiration? ). It is because advertising bombards people with technical terms witch they won't understand--on purpose. As to confuse people, while making them feel competent to actually decide to: buy their product. To really explain needs a few college level lectures in basic mathematics and physics. It is not possible to learn about such things in the context of loudspeakers alone. Sorry!

Dude, you are definitely "out there".
 

GelbeMusik

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Dude, you are definitely "out there".

Depends on the perspective, who is "out there", dude.

As best as I could tell, what that Mr. Brunhaver is saying ... The impression I'm left with is that he doesn't understand the reason that with most speakers ... done toward achieving this goal, and it makes perfect sense, irrespective of the fact that ...

This Mr Brunhave, self acclaimed Senior Speaker Designer by profession , aims to help You out of the audiophile confusion. You don't seem to appreciate his effort too much.
 

Another Bob

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So just so I’m clear on this, a properly designed passive crossover counld be designed with time alignment in mind? Not with a stepped or sloping baffle?
It is generally not practical (although perhaps theoretically possible) to perform time alignment using a passive crossover. But the driver delays are used as input to the crossover design process so that the result sums to a flat response through the crossover region. This is what JA is referring to in his Stereophile tests when he says things like "The decay of the tweeter's step blends smoothly with the start of the midrange unit's woofer step, which suggests optimal crossover design." (Quote from the review of the Von Schweikert Ultra 55 in the latest issue.) My main point was that taking a speaker and moving the drivers around without also changing the crossover is not a valid test of the audibility of time alignment, because it changes the frequency response.
 

KaiserSoze

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Depends on the perspective, who is "out there", dude.

This Mr Brunhave, self acclaimed Senior Speaker Designer by profession , aims to help You out of the audiophile confusion. You don't seem to appreciate his effort too much.

Dude, it is obvious to me that you have some sort of problem. I suppose I need to look back at all my posts and find the first one where you replied to one them, and maybe I'll be able to figure out what I did to have deserved the infatuation you have with me. I am inclined to ask you to STICK TO CRITICISMS OF WHAT SOMEONE HAS SAID, and to REFRAIN FROM PERSONAL ATTACKS, EVEN WHEN YOU WORD THEM IN A SICKLY OBTUSE MANNER. However I don't think there's a chance in hell that you would be able to do that in any remotely genuine manner. You would write something so bizarrely obtuse that nobody but you would have the first clue what the hell you were trying to say. But even though it would be a waste of time, I'm going to ask you anyway to do just that. You should have the decency and the intellectual maturity to explain, in a polite and civil manner, why you didn't like my comment on the video. You most likely did not even watch the video yourself, because for your purposes, watching the video and knowing what was in it would be of no value. All that mattered to you was the flimsy excuse to insult someone you don't know.

The comment I made about that video was on the mark. It was an informed, to-the-point criticism of what the guy in the video said. His remarks in the video clearly demonstrated confusion as to the purpose for vertical alignment of the drivers. He said in essence that it doesn't matter because there will still be an overall rotation in phase, thereby a lack of constancy in the time domain, from low to high frequency. This IS what he said in effect, i.e., it is what the whole amounted to, in total. Any anyone who knows the first thing about crossovers knows that this is wrong, i.e., that vertical alignment of the drivers is a very useful means by which to satisfy a requirement that must be satisfied in order for a crossover to work the way it is intended to work.

You remind me of that car in the car insurance commercial, you know, the fellow whose sole purpose seems to be for creating mayhem. In fact, I think you should change your ID here to "The Mayhem Dude". It would be perfectly appropriate.
 

KaiserSoze

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It is generally not practical (although perhaps theoretically possible) to perform time alignment using a passive crossover. But the driver delays are used as input to the crossover design process so that the result sums to a flat response through the crossover region. This is what JA is referring to in his Stereophile tests when he says things like "The decay of the tweeter's step blends smoothly with the start of the midrange unit's woofer step, which suggests optimal crossover design." (Quote from the review of the Von Schweikert Ultra 55 in the latest issue.) My main point was that taking a speaker and moving the drivers around without also changing the crossover is not a valid test of the audibility of time alignment, because it changes the frequency response.

Interesting. While the time misalignment, when the tweeter is mounted proud of the back of the woofer cone, is the same for all frequencies in the overlap region, the phase offset is clearly different for different wavelengths. I.e., a time-distance offset that corresponds to 40 degrees for one wavelength will correspond to 80 degrees at frequency double the first frequency. This raises the question of whether the phase shift introduced by an all-pass filter is constant irrespective of the wavelength. I'm not the least bit sure, but I suspect that it probably is, and if this happens to be correct, then it would seem that this type of simple network is perfectly capable of compensating for the time offset that is due to vertical misalignment of the drivers. But if so, i.e., even if it happens that the all-pass network will fully and ideally compensate for the time offset, then I still like the approach of tilting the baffle back.
 

Lorenzo74

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Hi Friends,

I have manged to directly experience on/off time alignment vs time coherent in my setup keeping all other things (almost) equal.
unfortunately you need Dirac live stereo version or DEQX (or rephase,...) I used Dirac Live stereo and minidsp.

This is what I did:

go through all measuring step until Dirac Live suggest you the corrected curve.
then manually move cursors up and down to match as much as you can the original speaker AMPLITUDE response, then save it.
then listen to different type of music playing with Dirac button adding and removing Dirac correction that in this specific case will be only about phase--> Dirac will only change phase keeping same amplitude response of your original not time aligned passive crossover speakers.
enjoy subjective impact of time alignment in your brain audio reconstruction process.

if you accept suggestions:
-Dave Brubeck Quartet, Take Five, 2:30-->3:30, drum attack...
-TOTO, I Will Remember, intro, first 10 seconds
-Johnny Cash, The Man Comes Around 2:05-->2:20

if you have doubts or seek for objective evidence just use REW to verify what I wrote. (be clement and accept minor amplitude differences, no one is perfect on earth )

@Flak , che ne pensi? può funzionare?

Note:
this doesn't work at all with speakers like those below:
https://www.stereophile.com/content/dunlavy-audio-laboratories-sc-iv-loudspeaker-measurements
since "This speaker is more like an electrical component in its phase performance. " JA 2008
or this
https://www.stereophile.com/content/vandersteen-model-seven-loudspeaker-measurements
or both of the following:
https://www.stereophile.com/content/kii-audio-three-loudspeaker-measurements
https://www.stereophile.com/content/dutch-dutch-8c-active-loudspeaker-system-measurements Wow...

looking forward to hear from you about your subjective feedback of this experiment.
my Best
L.
 

QMuse

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Hi Friends,

I have manged to directly experience on/off time alignment vs time coherent in my setup keeping all other things (almost) equal.
unfortunately you need Dirac live stereo version or DEQX (or rephase,...) I used Dirac Live stereo and minidsp.

This is what I did:

go through all measuring step until Dirac Live suggest you the corrected curve.
then manually move cursors up and down to match as much as you can the original speaker AMPLITUDE response, then save it.
then listen to different type of music playing with Dirac button adding and removing Dirac correction that in this specific case will be only about phase--> Dirac will only change phase keeping same amplitude response of your original not time aligned passive crossover speakers.
enjoy subjective impact of time alignment in your brain audio reconstruction process.

if you accept suggestions:
-Dave Brubeck Quartet, Take Five, 2:30-->3:30, drum attack...
-TOTO, I Will Remember, intro, first 10 seconds
-Johnny Cash, The Man Comes Around 2:05-->2:20

if you have doubts or seek for objective evidence just use REW to verify what I wrote. (be clement and accept minor amplitude differences, no one is perfect on earth )

@Flak , che ne pensi? può funzionare?

Note:
this doesn't work at all with speakers like those below:
https://www.stereophile.com/content/dunlavy-audio-laboratories-sc-iv-loudspeaker-measurements
since "This speaker is more like an electrical component in its phase performance. " JA 2008
or this
https://www.stereophile.com/content/vandersteen-model-seven-loudspeaker-measurements
or both of the following:
https://www.stereophile.com/content/kii-audio-three-loudspeaker-measurements
https://www.stereophile.com/content/dutch-dutch-8c-active-loudspeaker-system-measurements Wow...

looking forward to hear from you about your subjective feedback of this experiment.
my Best
L.

Yes, your experiment would work, but as I did my filters mannually with rePhase I was actually able to do a better thing: I can easilly generate filters with and without phase correction while keeping frequency response correction active. My experience is that audibilty of phase correction depends on the ammount of reflections in your room - the less reflections you have the better are the chances you will be able to notice the difference with phase correction. As my room is pretty rich with reflections I can't say I noticed the difference but I have tried the same thing in friends room that is heavilly treated and there we were able to notice the difference. But be aware it is very subtle and that correcting properly frequency response is much (and here I really mean much) more important.
 

tuga

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Yes, your experiment would work, but as I did my filters mannually with rePhase I was actually able to do a better thing: I can easilly generate filters with and without phase correction while keeping frequency response correction active. My experience is that audibilty of phase correction depends on the ammount of reflections in your room - the less reflections you have the better are the chances you will be able to notice the difference with phase correction. As my room is pretty rich with reflections I can't say I noticed the difference but I have tried the same thing in friends room that is heavilly treated and there we were able to notice the difference. But be aware it is very subtle and that correcting properly frequency response is much (and here I really mean much) more important.

This masking that you mention regarding untreated rooms helps to substantiate the claim that room reflections above transition produce (euphonic) distortion.
 

QMuse

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This masking that you mention regarding untreated rooms helps to substantiate the claim that room reflections above transition produce (euphonic) distortion.

I wouldn't really call it distortion as we are not perceiving reflections as a distortion. Every closed space has it's own acoustic signature that will be added to any sound source in that space, be it loudspeaker, live human voices, clapping hands sound or any other sound produced/played in that space (room).

What kind of acoustic signature you prefer is entirely subjective and definitely a personal choice. The same is with what kind of directivity you prefer with speakers, but as long as reflected responses are decently smooth speaker is doing a good job and it is up to you to like one more than the other.
 
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Frank Dernie

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I still believe we listen "through" rooms.
I have compared room compensation with non, my speakers were positioned to minimally excite room modes though and without the correction my bass is less exagerated than some people choose from their correction device :), Amir's review of Audessey for example.
My daughter is an accomplished cellist and I have never felt the need to correct her cello before listening, though a microphone shows the modes clearly and moving the microphone gives quite different recorded sound. I assume as people we quickly assimilate the acoustic fingerprint on a space we go into and compensate for it in our heads.
The recorder can't do this, of course, so the sound it picks up shows the room effect I don't notice.
Maybe I am unique/lucky but I definitely "see" the modes using a microphone and the sound of a recording sounds different with different microphone positions but I don't hear the difference in room, only on the recording.
This is also true in our piano room.
Our piano, a Steinway Model B, sound fine wherever one listens to it but record it and the sound you hear on playback changes with microphone position. It does not with ear position.

I first noticed the effect making live recordings decades ago.
IMO using a microphone output as analagous to hearing is a common mistake and quite wrong.
 

tuga

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I still believe we listen "through" rooms.
I have compared room compensation with non, my speakers were positioned to minimally excite room modes though and without the correction my bass is less exagerated than some people choose from their correction device :), Amir's review of Audessey for example.
My daughter is an accomplished cellist and I have never felt the need to correct her cello before listening, though a microphone shows the modes clearly and moving the microphone gives quite different recorded sound. I assume as people we quickly assimilate the acoustic fingerprint on a space we go into and compensate for it in our heads.
The recorder can't do this, of course, so the sound it picks up shows the room effect I don't notice.
Maybe I am unique/lucky but I definitely "see" the modes using a microphone and the sound of a recording sounds different with different microphone positions but I don't hear the difference in room, only on the recording.
This is also true in our piano room.
Our piano, a Steinway Model B, sound fine wherever one listens to it but record it and the sound you hear on playback changes with microphone position. It does not with ear position.

I first noticed the effect making live recordings decades ago.
IMO using a microphone output as analagous to hearing is a common mistake and quite wrong.

I was referring to room "interference" above the transition frequency though, boundary reflections in the midrange and treble.
 

QMuse

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I still believe we listen "through" rooms.
I have compared room compensation with non, my speakers were positioned to minimally excite room modes though and without the correction my bass is less exagerated than some people choose from their correction device :), Amir's review of Audessey for example.
My daughter is an accomplished cellist and I have never felt the need to correct her cello before listening, though a microphone shows the modes clearly and moving the microphone gives quite different recorded sound. I assume as people we quickly assimilate the acoustic fingerprint on a space we go into and compensate for it in our heads.
The recorder can't do this, of course, so the sound it picks up shows the room effect I don't notice.
Maybe I am unique/lucky but I definitely "see" the modes using a microphone and the sound of a recording sounds different with different microphone positions but I don't hear the difference in room, only on the recording.
This is also true in our piano room.
Our piano, a Steinway Model B, sound fine wherever one listens to it but record it and the sound you hear on playback changes with microphone position. It does not with ear position.

I first noticed the effect making live recordings decades ago.
IMO using a microphone output as analagous to hearing is a common mistake and quite wrong.

IMO it is not about ears vs microphone, it is about our perception of live play and recording been played by speakers.

Besides, your daughter will immediately adapt when she plays a tone which hits a peak or a dip caused by the room and she will adjust accordingly for the tone to be played at right loudness while speakers playing a recording will not, so they need to be EQ-ed.
 

tuga

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I wouldn't really call it distortion as we are not perceiving reflections as a distortion. Every closed space has it's own acoustic signature that will be added to any sound source in that space, be it loudspeaker, live human voices, clapping hands sound or any other sound produced/played in that space (room).

What kind of acoustic signature you prefer is entirely subjective and definitely a personal choice. The same is with what kind of directivity you prefer with speakers, but as long as reflected responses are decently smooth speaker is doing a good job and it is up to you to like one more than the other.

This may be fine with pop and rock studio recordings but with recordings which already have an acoustic signature (mostly minimally-mic'ed classical music) the more "interference" from the listening room the less one is able to listen to the cues of the original venue. It is a form of distortion, like when singing in a bathroom or in a church the sound becomes more diffuse / less focused and the imperfections less obvious.

I agree that ultimately its all down to personal preference but I don't think it makes sense to criticise tailored frequency response (i.e. BBC dip, etc.) or a pinch of low- even-order harmonic distortion and condone wide-directivity speakers or a more reflective room.
 
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