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step response is a important part to show speed of speaker that is good enough for ITD. See measures

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KSTR

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when the reason for good stereo wide is not a fast step response, what reason do you think it is and with which measure value can see a good stereo width. what do you think ?
You can correct the step response of a speaker with DSP (phase-correction of the input signal) so that is is perfect: miminum phase, strictly following the frequency response.

When you do that you will find speakers that had different perceived soundstage before will still image as different as they did, after correction. Overall, imaging improves a little bit in both cases, though.

Actually a simpler test is using a good fullrange driver and add excess phase as if there were actually an XO in place. Again, imaging characteristics don't change very much.

Conclusion: Step Response shape (or more correctly, excess phase contribution) has some impact on imaging but it is not the key parameter.
 
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bennybbbx

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I think it is desirable to have a more linear time response, for sure. But I highly doubt one would automatically get the impression of better "stereo width" perception by just that one property. I could, for instance, physically position speakers closer or farther apart to change the perceived stereo separation.

I mean for nearfield on desktop. I use 80 cm .and i think i find the best test for speaker driver speed. maybe i plug a headphone that have 8 ohms instead the kali woofer and measure this. then can see how much the speaker slow it down or the filter they use.

or what happen when i connect to the class d amp a 30 ohm headphone.does this this work good enough for test
 
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bennybbbx

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You can correct the step response of a speaker with DSP (phase-correction of the input signal) so that is is perfect: miminum phase, strictly following the frequency response.

When you do that you will find speakers that had different perceived soundstage before will still image as different as they did, after correction. Overall, imaging improves a little bit in both cases, though.

Actually a simpler test is using a good fullrange driver and add excess phase as if there were actually an XO in place. Again, imaging characteristics don't change very much.

Conclusion: Step Response shape (or more correctly, excess phase contribution) has some impact on imaging but it is not the key parameter.

and what do you think is the key parameter for stereo width and depth of field ?. the step response can correct but it is impossible to correct the phase in the crossover range. because the phase between tweeter and wooferin crossover range is always fixed. if you do shifts then tweeter and woofer move.
 

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^ Simple but probably unsatisfying answer: everything matters in some way or another, plus judging imaging properties is a very personal thing.
I've never found a clear recipe what exactly creates the soundstage illusion that I prefer, but good dispersion, good FR, good individual driver phase tracking, and good step response do matter. Pet theory would be that the less work the brain has to "hear through" artifacts the easier we can perceive a wide and deep soundstage.

With many recording soundstage also changes when the signal polarity is flipped.

Stereo illusion is really fragile simply by the crude concept of 2-speaker stereo.
 

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and what do you think is the key parameter for stereo width and depth of field ?. the step response can correct but it is impossible to correct the phase in the crossover range. because the phase between tweeter and wooferin crossover range is always fixed. if you do shifts then tweeter and woofer move.
No offense meant but you seem to be unaware of some important facts surrounding acoustics, psychoacoustics and recording techniques. There's a whole world waiting for you to explore. I recommend these 2 books to get you started:

Howard/Angus "Acoustics and psychoacoustics"
Toole "Sound reproduction"

I find this one of the best basic articles ever written on stereo (recording): http://decoy.iki.fi/dsound/ambisonic/motherlode/source/Stereo microphone techniques_arethe purists wrong_Lispshitz_1986_pt1.pdf

Good luck on your journey and have fun!
 
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bennybbbx

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No offense meant but you seem to be unaware of some important facts surrounding acoustics, psychoacoustics and recording techniques. There's a whole world waiting for you to explore. I recommend these 2 books to get you started:

Howard/Angus "Acoustics and psychoacoustics"
Toole "Sound reproduction"

Good luck on your journey and have fun!

I know that the common opinion is that the beam width of the tweeter should bring better depth of field and stereo width. this stand in such books. but i have my speakers 50 cm near and i can hear good depth of field and big concert halls. i dont think that wider directivity can bring out of my 17 qm room the reverb of a concert hall. it come only from speaker and wide directivity help nothing more as bring more reverb of a 17 qm room. also i have done a video example that make all frequency below 460 hz mono. the area that all speakers have wide directivity. you need be able to hear ITD. when make this range mono the stereo width get very few and few depth of field. so or only for my ears this range upto 460 is also very important for stereo width and depth of field.
 
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ctrl

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EDIT: here is a overlay compare between the FR of kali and the much EQ headphone. when use less smothing can see kali do from 1.5 khz to 2 khz -20 db. thats a very high order filter.
Why do you smooth the measurements at all, there must be no room resonances in them, otherwise the comparisons are completely meaningless.
Why does the headphone measurement suddenly stop at 45Hz and the Kali measurement goes down to 30Hz?
1638385101389.png

Have you measured the headphones so that it is halfway realistic, either with a dummy head or at least the ear cup covered with a plate (plus felt) and the mic fed through a small hole?

If you perform such a measurement correctly, the frequency response will look like this, for example:
Beyerdynamic DT990
1638385708284.png

The evaluation of the step response according to your voodoo method (20% to 20% amplitude) then looks like this:
1638385732919.png
:eek:OMG, the DT990 headphone needs 5000µs (see gate length, 20% to 20% amplitude), which is four times slower than the Kali LP6. The headphones are so slow that you fall asleep while listening.

But you were also told that in the last thread.

How did you improve your measurements, are room resonances excluded in your measurements?
 
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bennybbbx

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Why do you smooth the measurements at all, there must be no room resonances in them, otherwise the comparisons are completely meaningless.
Why does the headphone measurement suddenly stop at 45Hz and the Kali measurement goes down to 30Hz?
View attachment 169426

i am 1.5 cm from driver away. if room make a problem i have verify when i have move microphone 3 cm away. step response is the same. and on superlux also the bass go very low with high level. the kali measure is the black line and show values at lower frequency, because kali is smooth 1/12 you can see this in the screenshot below. REW show smoothing value . the smooth of the headphone was 1/3. REW start at later frequency because of more smoothing. the kali measure is older. in my first post you see you can see the date and measure range. i have no head for headphones. i just
do it this way. :D
P3920648_2.JPG
the kali is just so slow that it doesnt matter.

maybe you can try out a step response with REW. maybe arta is diffrent. it depend also on hold time of measure signal. it look arta use a too long hold time. maybe rew use a short hold time. i have measure many headphones. the step response look always fast. here another headphone
panasonic rpth 030.jpg




The mems microphone from arc 2.5 is not omnidirectional. so it give less noise from behind and side. i have calibration file for this.
Have you measured the headphones so that it is halfway realistic, either with a dummy head or at least the ear cup covered with a plate (plus felt) and the mic fed through a small hole?

If you perform such a measurement correctly, the frequency response will look like this, for example:
Beyerdynamic DT990
View attachment 169431

The evaluation of the step response according to your voodoo method (20% to 20% amplitude) then looks like this:
View attachment 169432

see my first post of the headphone without filter. do you use a correction filter for headphone ?

:eek:OMG, the DT990 headphone needs 5000µs (see gate length, 20% to 20% amplitude), which is four times slower than the Kali LP6. The headphones are so slow that you fall asleep while listening.

But you were also told that in the last thread.

How did you improve your measurements, are room resonances excluded in your measurements?

you can also try out and create in REW a impulse response for a 1.5 khz 12 db filter export as wav. then import this wav impulse response and look step response of it. this is too around only 300 µsec long.
 
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dominikz

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1.5 khz high shelf 48 db.jpg


does for others that can hear ITD 48 db filters sound good ?. I am really surprise that 48 db filters give so slow step response.
A low-pass filter removes high frequencies so by definition will increase rise-time of a step response. You need high frequencies to be able to 'fit' sharp rising waveforms. Linear system theory and Fourier transformation calculations describe this completely.

It is also easy to illustrate this with REW, here's an example of a ideal pulse (red - ideally flat FR from 0-96kHz) and the same signal low-passed at various frequencies (note that for simplicity I'm only showing magnitude and not phase):
Ideal step (96kHz BW) vs the same ideal signal with steep low-pass filters at various frequenc...jpg

Here's the same responses in the time domain (step response):
Ideal step (96kHz BW) vs the same ideal signal with steep low-pass filters at various frequenc...jpg

Notice how, the more high frequencies you take away with a low-pass filter, the more time it takes for the step to reach full amplitude (and also there is more ringing before it stabilizes).

On the other hand removing low frequencies with high-pass filters will not affect rise time (which will remain 'fast', if you like), but will change the shape of the step function after the initial rise (since you need very low frequencies to 'fit' functions with constant values such as a step function):
Ideal step (96kHz BW) vs the same ideal signal with steep high-pass filters at various frequen...jpg
Ideal step (96kHz BW) vs the same ideal signal with steep high-pass filters at various frequen...jpg


Of course humans don't hear frequencies below ~15-20Hz nor those higher than about 20kHz, so a non-ideal step signal sounds exactly the same as an ideal one to us :):
Ideal step (96kHz BW) vs the same ideal signal band-limited to human hearing range - Frequency...jpg

Same signals in time domain (zoomed-in to better see the rise-edge):
Ideal step (96kHz BW) vs the same ideal signal band-limited to human hearing range - Time domain.jpg

Same signal but zoomed out to better see the overall function shape:
Ideal step (96kHz BW) vs the same ideal signal band-limited to human hearing range - Time doma...jpg


In short, the step response of any audio device that can be approximated as a linear, time-invariant (LTI) system, which loudspeakers normally are, is completely described by the frequency magnitude (+phase) response. I'd suggest to do some reading on linear systems theory and Fourier analysis/transformation, as they are a very helpful tool to gain a deeper understand of audio (and some other things as well :)).

Hope this is helpful! :)
 

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i have no head for headphones. i just
do it this way. :D
I had thought something like this, but to measure like this is unfortunately not quite optimal (the formulation for people from the Anglo-Saxon language area) or, to put it harshly, total nonsense (for the German-speaking area) ;)
1638397226978.png

Didn't you notice that you lose over 30dB of sound pressure level in the bass range compared to the treble when you measure your headphones like this.
1638397843740.png

What does this mean for the step response?
After all these months from the last thread, this should be clear.

Here's an example that's similar to your headphone versus speaker comparison.

In blue you can see the frequency response of an ideal, typical loudspeaker with an early high frequency drop and decent low frequency response.
In brown you see the response of your headphone measurement (the difference between highs and lows is only 20dB).
1638397982343.png

What this means for the step response has already been explained to you several times in the last thread:
1638398345231.png 1638398364293.png

Update: ...and by @dominikz before my post (thanks for that).
 
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bennybbbx

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and wy have the ideal step response a decay of 2,8 ms to reach 0 ?. I use speaker correction can correct my headphone with EQ phase linear and i never get such a step response. I correct the headphone with linear phase and nearest phase filters to simulate a LP with -36 db. sure all filters change the step response.

but look the step response and the frequency response of meaasured headphone. FR is near same but the linear phase step response look lots faster. and this is what i hear . speakers that look faster sound better in stereo width.
686 non phase linear.jpg


686 phase linear.jpg


the EQ settings i use. you can see the LP filter begin from 1.2 khz. this cause a longer time in compare to 1.5 khz. but still much faster as the Kali. and to make it clear. I use the Eq before the signal go to headphone. the left value scale that show at lowest -36 is the value that is set in the EQ. range stand at 40 db. the range knob can change so i can set the lowest limit. for example when i set range to 60 db then the lowest is -60 db

the eq setting.jpg


overlay from linear phase EQ and non línear phase EQ 1/48 smoothing

fr compare 686 linear phase to nearest.jpg



@ctrl

and if you think that this large step response diffrence with very simular FR are not possible, i can also do a video how i measure and switch between phase linear and not phase linear. or should i do a live stream ?.
 
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dominikz

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and wy have the ideal step response a decay of 2,8 ms to reach 0 ?.
Sorry, but where do you see this? If you're talking about this diagram:
index.php

The red plot is the ideal step function (though limited to 96kHz bandwidth), and you see it rises from 0-100% immediately. The purple plot gets to 0-100% in about 2,4ms, but this is the ideal step response filtered with an 8th order Linkwitz-Riley low-pass filter at only 100Hz (that might be similar to e.g. a subwoofer response - though the LP order is higher in my example above than you would normally use with a sub).

I correct the headphone with linear phase and nearest phase filters to simulate a LP with -36 db. sure all filters change the step response.
and if you think that this large step response diffrence with very simular FR are not possible, i can also do a video how i measure and switch between phase linear and not phase linear. or should i do a live stream ?.
If you change the phase-response of the filter the resulting step responses will look different as well - sure. The magnitude vs frequency response in that case may look the same, but the phase vs frequency response will be changed - this is of course predicted by the theory as well. Note that the 'frequency response' of a system means magnitude + phase response vs frequency, not just magnitude.

but look the step response and the frequency response of meaasured headphone. FR is near same but the linear phase step response look lots faster. and this is what i hear . speakers that look faster sound better in stereo width.
686 non phase linear.jpg


686 phase linear.jpg


the EQ settings i use

the eq setting.jpg


overlay from linear phase EQ and non línear phase EQ 1/48 smoothing

fr compare 686 linear phase to nearest.jpg
Note that it is possible to achieve nicer ('faster'?) looking step responses of loudspeakers as well if you use phase correction filters.
E.g. Dirac Live does this automagically (from this thread); first let me show the correction filter it applies (ignore the fuzz in HF, it is a measurement artefact):
index.php

In blue below is the Dirac Live corrected loudspeaker response (using above filter), and in black is the original loudspeaker step response without any EQ:
index.php

You can see that the all-pass filter around 2,1kHz is used to align woofer and tweeter phase, which results in a nicer-looking step response.

My personal experience with phase corrections (using Dirac Live, Audiolense and rePhase) has so far been inconclusive; I'm not really convinced that under controlled and blinded conditions I could reliably distinguish between the same magnitude filter on the same system/environment with and without phase/time correction (excluding some huge phase errors, of course). This would IMHO require an ABX to determine.
 
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bennybbbx

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Sorry, but where do you see this? If you're talking about this diagram:

I mean this diagram and i have add the lines of the 2.8 ms i mean.
step.jpg


and thanks for posting the JBL 305 step response. the without DRC left measure is without any EQ ?. it look slower but in compare to kali the time the level have around 90% look much fewer on your step response. maybe this is bad when the step response is much curved as from the Kali and have no sharp edge.

but it is still strange wy i get such a much smaller step response with the headphone correctet and LP 1.2 khz. the FR look very linear from bass until crossover freq come.

I have done hear test with linear filter EQ -30 db 1.5 khz. with speaker i hear many diffrence in stereo width. but not so less as with the Kali. with headphone not so much. the not linear phase sound smaller. i never hear before a diffrence between linear phase or not. but with this high order filter i hear this much. i upload soon a hear test. maybe other can hear the diffrence too

EDIT: i have now do a overlay of the kali step response and the headphone with EQ correct minimal phase and phase linear. klai is much slower in raise time too, but i use for the headphone 1.2 khz LP. normaly it should slower. but the kali have 400 µsec raise time. the raise time of headphone is 200 µsec. this mean the kali have 2 * slower raise time.
step compare.jpg
 
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KSTR

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I mean this diagram and i have add the lines of the 2.8 ms i mean.
The lower (== better) the frequency response is going the longer it takes. That means the tail -- tells you nothing about "speed", only about low frequency response (as it is the same data just displayed differently).

Also note that the tail always must go through zero and then below, before finally settling slowly at 0. A speaker cannot put out DC sound pressure and that'sd why the integral (area under the curve) of the step response must be zero. The order of the low-frequency roll-off and the Q factor determine the shape of the oscillation of the step response.

I guess all this is probably way above your head.
 
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bennybbbx

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I guess all this is probably way above your head.

i have edit my last post and add a overlay of the step responses. have you see this and what do you think is the reason wy the raise time of kali is 2* slower but the headphone use 1.2 khz 30 db lowpass ?. the kali have 1.5 khz lowpass and i guess a 48 db shelving filter that reduce after 2 khz only 22 db until end. a lower LP frequency increase raise time. so my conclusion is driver of kali is very slow and step response show it clear
 
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I think that when an understanding of the actual relationship between bandwidth and impulse response shows up, I'll consider commenting. Ditto a good understanding of linear superposition

I understand Linear superposition and i undersand what you want say. frequency and driver have influence of the step response. I agree to this. But this is no forumula with 2 unknown parameter because the Frequency response is measure too and a software can calc step response with normalize FR to better compare. or just use a 100 hz 24 db highpass for all speakers, so the bass range is not so low. I understand that in Bass range small and large speakers are much diffrent in FR. this have a little influence of the fall time of the step response.

here you can see much diffrent FR of the JBL done with EQ in Bass Range. but the fall/rise time from 20% to 20% level increase only a little from 580 µsec to 650 µsec (this are 12%) with this much diffrent Bass frequency change. the kali step response 20% to 20% time need 1440 µsec but have not so much bass as the JBL with the EQ. so because of the kali bass there need much shorter step response, and so conclusion is the driver is very slow.
EDIT: the JBL measure in compare at 1.5 cm. because the room have very less influence it can give much bass. because its coaxial the tweeter is see too in FR and step response
step response compare 1.5 cm.jpg
frequency compare 1.5 cm.jpg


so i think the step response of a speaker is important to see. if you like it more, then can ignore the fall time and show only the raise time. impulse response i think not usefull when test complete speakers. because in a impulse response it is hard to see which is the raise and fall time of the mid/bass. or maybe need see the impulse response of bass/mid or mid only to see if it is good for ITD
impulse compare.jpg


Or can somebbody explain how i can see on the JBL and kali and headphone impulse response compare which is faster ?????. i can only see the kali look slow. but the headphone look much slower as jbl but it is not
 

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I understand Linear superposition and i undersand what you want say. frequency and driver have influence of the step response. I agree to this. But this is no forumula with 2 unknown parameter because the Frequency response is measure too and a software can calc step response with normalize FR to better compare. or just use a 100 hz 24 db highpass for all speakers, so the bass range is not so low. I understand that in Bass range small and large speakers are much diffrent in FR. this have a little influence of the fall time of the step response.

here you can see much diffrent FR of the JBL done with EQ in Bass Range. but the fall/rise time from 20% to 20% level increase only a little from 580 µsec to 650 µsec (this are 12%) with this much diffrent Bass frequency change. the kali step response 20% to 20% time need 1440 µsec but have not so much bass as the JBL with the EQ. so because of the kali bass there need much shorter step response, and so conclusion is the driver is very slow.
EDIT: the JBL measure in compare at 1.5 cm. because the room have very less influence it can give much bass. because its coaxial the tweeter is see too in FR and step response
View attachment 169839View attachment 169840

so i think the step response of a speaker is important to see. if you like it more, then can ignore the fall time and show only the raise time. impulse response i think not usefull when test complete speakers. because in a impulse response it is hard to see which is the raise and fall time of the mid/bass. or maybe need see the impulse response of bass/mid or mid only to see if it is good for ITD
View attachment 169838

Or can somebbody explain how i can see on the JBL and kali and headphone impulse response compare which is faster ?????. i can only see the kali look slow. but the headphone look much slower as jbl but it is not

Pardon my ignorance but why are you so hung up on this "fast/slow" thingy? Any speaker is "fast" enough if it can play back signals within the range of human hearing. A "slow" speaker would simply manifest itself in a bandwidth-limited response.
I think right now you're looking at a completely unrelated topic when you really want to know more about stereo and human sound perception. The field has a name: "psychoacoustics".
It was pioneered in the 70s by Jens Blauert and others. There's tons of literature in German. Also look for Fastl/Zwicker.
 
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Pardon my ignorance but why are you so hung up on this "fast/slow" thingy? Any speaker is "fast" enough if it can play back signals within the range of human hearing. A "slow" speaker would simply manifest itself in a bandwidth-limited response.
I think right now you're looking at a completely unrelated topic when you really want to know more about stereo and human sound perception. The field has a name: "psychoacoustics".
It was pioneered in the 70s by Jens Blauert and others. There's tons of literature in German. Also look for Fastl/Zwicker.

it is because of Interaural time delay(ITD). some(i dont know how much) less or more Humans can hear position diffrences in phase of 40 µsec at least. this is no hifi voodoo. a angle of 10 degree is hear when 1 channel is delay with 0.08 ms or 80 µsec. the brain can create out of this delay a room feeling from the reverb reflections that are record in the music. many people like sound in a concert hall more as in a bathroom or small room. and near all pop songs use reverb too. A bass/mid woofer that is only able to produce 5 khz (which is 200 µsec period time) can not produce the signal phase exact. left and right speaker play always at same time diffrent membrane movements and diffrent frequency. so it is physics. a speaker that can only play signals at max period time of 200 µsec with -3 db can never be able to play signals at lower frequency phase exact for 40 µsec. so a fast system is need. i think the best speaker is when use a fullrange speaker from bass to mid that work upto 20 khz and use a tweeter. or look at the step response of a wide band speaker. it is much faster also because it need no crossover. and many say wide band speakers have best stereo width and there can still buy the old 1960 avatone mixcubes for a high price. they sound very good in stereo. you can see and read in my ITD hear test video, what diffrence can hear when delay the signal from 20 µsec to 200 µsec (0.2 ms) . i can hear ITD and i hear lots diffrence and slow speaker do sound not good. ITD is usefull when hear at lower volumes too, to have a great sound. when i make the kali very loud it think it sound ok and the effect is that the level diffrences when hear loud are larger between left and right so the brain can better produce room feeling. but hear loud is bad for ears so fast speakers that sound awesome at low level are good for ear health

from my video text. and btw now youtube disable the dislikes. i see some videos by others that show dislikes. does somebody know how i can switch it on ?

https://www.sciencedirect.com/topics/medicine-and-dentistry/binaural-hearing "Binaural hearing uses two main acoustic cues: interaural time difference (ITD) and interaural level difference (ILD) (Doerbecker and Ernst, 1996; Francart et al., 2011): • ITD is the delay between both ears. It is efficient for low frequencies (below 850 Hz). It is due to the envelope of the signal reaching the two ears. It can be reminded that a sound coming from the side at 90° has an ITD of 0.6 ms. When the source is situated in the front (azimuth 0°), the so-called front target, the ITD is 0 ms. • ILD is related to the intensity reaching the two ears. The signal is more or less attenuated by the head shadow. This effect is mostly perceptible with high frequencies (above 3 kHz). ILD is 0 for the front target." a period time of 1 ms is 1 khz frequency. humans are able in tests to hear 0.01- 0.02 miliseconds delay diffrence between speaker. thats less than 1/ 44.1 khz sample here is delay to angle diagram and also a test audio example https://www.sfu.ca/sonic-studio-webdav/handbook/Binaural_Hearing.html



EDIT: maybe this example explain it. a old VHS tape video recorder can play 400 dark white lines and it have some jitter. now when play a large black white block you see the jitter but the frequency is not high. and same happen with slow speakers in stereo. they jitter and wobble alot and this cause random jitter also known as phase changes.



EDIT: and on this sound editor(there is a image in the link) you can see left and right channel overlayed and can see the left and right channel do much diffrent movements of the mid range. sometimes a channel do nothing sometimes do most. did you really think a speaker that can only play period time of 200 µsec(5 khz) -3 db is able to reach between left and right channel the positon of the source signal exact for at leat 40 µsec ?

 
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it is because of Interaural time delay(ITD). some(i dont know how much) less or more Humans can hear position diffrences in phase of 40 µsec at least. this is no hifi voodoo. a angle of 10 degree is hear when 1 channel is delay with 0.08 ms or 80 µsec. the brain can create out of this delay a room feeling from the reverb reflections that are record in the music. many people like sound in a concert hall more as in a bathroom or small room. and near all pop songs use reverb too. A bass/mid woofer that is only able to produce 5 khz (which is 200 µsec period time) can not produce the signal phase exact. left and right speaker play always at same time diffrent membrane movements and diffrent frequency. so it is physics. a speaker that can only play signals at max period time of 200 µsec with -3 db can never be able to play signals at lower frequency phase exact for 40 µsec. so a fast system is need. i think the best speaker is when use a fullrange speaker from bass to mid that work upto 20 khz and use a tweeter. or look at the step response of a wide band speaker. it is much faster also because it need no crossover. and many say wide band speakers have best stereo width and there can still buy the old 1960 avatone mixcubes for a high price. they sound very good in stereo. you can see and read in my ITD hear test video, what diffrence can hear when delay the signal from 20 µsec to 200 µsec (0.2 ms) . i can hear ITD and i hear lots diffrence and slow speaker do sound not good. ITD is usefull when hear at lower volumes too, to have a great sound. when i make the kali very loud it think it sound ok and the effect is that the level diffrences when hear loud are larger between left and right so the brain can better produce room feeling. but hear loud is bad for ears so fast speakers that sound awesome at low level are good for ear health

from my video text. and btw now youtube disable the dislikes. i see some videos by others that show dislikes. does somebody know how i can switch it on ?





EDIT: maybe this example explain it. a old VHS tape video recorder can play 400 dark white lines and it have some jitter. now when play a large black white block you see the jitter but the frequency is not high. and same happen with slow speakers in stereo. they jitter and wobble alot and this cause random jitter also known as phase changes.

You're mixing up completely different concepts. Interaural ("between the ear") time and phase differences are independent of how "fast" the waveform coming from the speaker actually is. By the way, the nerves in your ear can only fire at a rate of about 1ms. They are really "slow". How does that fit in your "theory"? Rest assured speakers are "fast" enough otherwise you won't be able to hear high frequencies from a speaker.
Please read the PDF I've linked above about stereo. It really touches every aspect. You'll even learn how ILD translate to ITD in stereo reproduction. ILD/ITD is important to localization but not to how you perceive spaciousness/envelopment/image width from a stereo recording (although there IS a connection depending how similar the response of two stereo speakers is, and, there can be "false" spaciousness from phase aberrations).

P.S. There are no "Avatones". They were called "Auratones". I've used those to check how a mix would translate to low quality car radios. They sound awful and that's why they were used ;)

P.P.S. My current speakers look like this. Not because they are "fast" but because they provide (rather) smooth and high directivity:
 

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bennybbbx

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You're mixing up two completely different concepts. Interaural ("between the ear") time and phase differences are independent of how "fast" the waveform coming

about phase stand in audio knowledge

https://blog.landr.com/out-of-phase-audio/ Phase in audio refers to the position of a sound wave in time. Think of a sine wave on a graph. The wave's phase is its position along the x-axis.
this i mean the position in x axis from left and right speaker in the frequency range from 150 hz upto 900 hz need be good at leat 40 µsec between left and right to have good stereo with ITD. and this is guess is also the reason that there are some 3 way speakers, because they have faster mid because it is smaller


from the speaker actually is. By the way, the nerves in your ear can only fire at a rate of about 1ms. How does that fit in your "theory"?

I dont know. 1 ms is only a frequency of 1 khz. and it is not clear in which time raster nerves can fire. for example some nerves can fire at 1.2 ms others fire on 1.202 ms. if it is totally clear how audio work then all speakers need sound good for people who hear ITD.


Rest assured speakers are fast enough otherwise you won't be able to hear high frequencies from a speaker.

the woofer/mid is with LP filter limit, but it seem not able to give with timeerror <= 40 µsec on X position the correct level from left and right speaker. it do jitter. the high freq is play with the tweeter from around 1.5 khz upto 20 khz. if the phase (x position) is exact or not on tweeter i think not important. i have ribbon tweeter test but it do not enhance stereo width.so it seem above 1.5 khz brain need only frequency and exact x position (or phase) does not matter


P.S. There are no "Avatones". They were called "Auratones". I've used those to check how a mix would translate to low quality car radios. They sound awful and that's why they were used ;)
I have read audio theory. i mean avantone and have only forget a n.
the auratones i not mean . i mean this http://www.avantonepro.com/mixcubes-active-creme.php they get good testresult for stereo width. maybe tests in german too. on bonedo the guy that test seem hear itd. he tell kali sound strange and jbl 104 sound very wide


P.P.S. My current speakers look like this. Not because they are "fast" but because they provide (rather) smooth and high directivity:

this is a subwoofer only or wideband speaker ?
 
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