Step Response: Does It Really Matter?

[bennybbbx]

@bennybbbx , you confuse the most basic concept of phase shift and confuse this with speed. Time delay(phase) is not speed. It is time delay. I suggest doing a lot more reading before trying to attack this subject. You are going around in a circle. Sorry but I don't have time for the necessary lesson.

you have not answer the question. you write about time delay. I mean time delay is the result of too slow speaker speed. Can say time delays(and overshoots amd undershoots) depend on speaker speed and level changes the speaker must do. this i mean. and the result is that a mid bass woofer that reach only 6 khz -3 db(which many big not better do) have problem to reach good enough time for ITD in correlation to the other speaker when the waveform level changes of left and right speaker is diffrent at same time(which happen near allways in stereo signals. confirmed in wave screenshots)

 

[bennybbbx]

I find a thread aboutm measure of speaker speed here. https://www.audiosciencereview.com/forum/index.php?threads/speaker-speed.17759/

you can read this

I didn’t read the thread, but there should be a clear conclusion!

Measure the impulse response (or any signal that can be transformed into an impulse response, eg a sweep). That measurement will contain a full characterisation of your loudspeaker’s transient response (within the limits of your measurement setup, ofc).

It’s really that simpl

 

[j_j]

Benny, also, please understand that a multidriver speaker system is not limited by the "speed" of its slowest element, but rather by its fastest element. I think there are things about linear summation and superposition that you aren't grasping, but I'm not sure.

 

[bennybbbx]

Benny, also, please understand that a multidriver speaker system is not limited by the "speed" of its slowest element, but rather by its fastest element. I think there are things about linear summation and superposition that you aren't grasping, but I'm not sure.

See my waveform screenshots of the frequency below 2 khz. this waveforms are not produce by the tweeter, but because humans can hear ITD the slow woofer mid do no good sound, because it can not produce the peaks for transients at the precision that is need for good ITD. < 0.05 ms at least. it vary alot on slow speakers. depend on how much it move

 

[j_j]

See my waveform screenshots of the frequency below 2 khz. this waveforms are not produce by the tweeter, but because humans can hear ITD the slow woofer mid do no good sound, because it can not produce the peaks for transients at the precision that is need for good ITD. < 0.05 ms at least. it vary alot on slow speakers. depend on how much it move

You do not understand linear summation. Please, go get some math, check out that Fourier Analysis book. You're repeating a mistake over and over.

Your comment on the "speed" of 2kHz is correct, but has very little to do with the problem.

 

[bennybbbx]

You do not understand linear summation. Please, go get some math, check out that Fourier Analysis book. You're repeating a mistake over and over.

Your comment on the "speed" of 2kHz is correct, but has very little to do with the problem.

there is no summation. In my ITD testvideo you can disable the tweeter. the tweeter do nothing, because it use low pass. the low freq woofer need create the correct waveforms for ITD. it must be precise to <0.1 ms and a slow wobbling overshooting undershooting woofer can not do that. wy is this not clear ?. i do not know exact what the problem is. maybe it is more additional the problem of overshoot undershoot of systems. the JBL have in step response much fewer overshoot undershoot as my other speakers.

if you think a woofer that can produce 3 khz is good enough for ITD, wy do you think there are 3 way systems(that are often expensive) with a additional mid range speaker ?. this need more crossover and more phase problems can happen. or wy many say that the avatone mix cubes that are full range speakers give better stereo image.

Also Kali have 3 way systems the in5 and in8.

maybe you should compare with images. a old VHS recorder do image tearing between line always. there come images in video, that have no high frequency. for example a blured rectangle. they too show tearing 1 or 2 pixels. see as left speaker is for odd lines right speaker for even lines. and for ITD the signals between such lines need exact. i know not so good to compare. but maybe it help to undertand wy ITD is a problem for slow 2 way systems

 

[audio2design]

There are so many things wrong with your last statement and ones before it, there is really no clear place to start @bennybbbx. You have been given good advice on upgrading your knowledge.

 

[bennybbbx]

There are so many things wrong with your last statement and ones before it, there is really no clear place to start @bennybbbx. You have been given good advice on upgrading your knowledge.

you give me no good advice. fft have nothing to do with the fact that for ITD even if the speaker need do only frequencies of 1-2 khz in max the precision of the output waveform need be exact to less 0.1 ms.

I dont believe you that a bass/mid speaker that reach only 6 khz can do this. maybe you record such a speaker at 1 cm distance a 1 khz peak and compare the waveform with the original waveform if there is no diffrence more than 0.1 ms in the peaks. if you do that you see that this waveform is not exact on slow speakers it have much more diffrence. the delay diffrence depend alot how much the speaker need move.

If you think this is on left and right speakers realy same delay try record a multitone 1 time 180 and 1500 hz other time 450 and 1500 hz. now do the 2 records overlay with the original and show me that the diffrence of peaks between is less than 0.1 ms of the recordet waveforms. when you can do that i believe it. i do tests with wave recordings 1 cm away and i notice that the 1. wavecycle have much lower level. you can do all measure 1 cm away from speaker so reverb of room have no influence. you can also put the microphone in damping material. room influence can see when a single frequency testtone have changes in amplitude. but you need not check for level diffrence in the multitone test you need only check for peak diffrence in time

Now i wait for your screenshots

 

[audio2design]

You will be waiting a very long time for those @bennybbbx. I would use that time wisely.

 

[j_j]

You will be waiting a very long time for those @bennybbbx. I would use that time wisely.

Start with linear superposition. Please.

 

[j_j]

I'll try this ONCE.

Here I have a plot of 3 impulse responses.

The first is 30-3000 Hz sampled at 96000
The second is 3000-20000 sampled at 96000 There is no overlap in frequency between the two plots.
The third is their sum, which in fact is an impulse of 30-20000

Now, this is a symmetric response, which you can come very close to getting out of a loudspeaker if you work hard and use the proper digital crossover, but in fact you'll see the same kind of result if you did this with a minimum phase filter (i.e. high level crossover with r's and c's and l's) tht was done correctly, except that the peak would come first. The summation would still result in similar results.

Note, there are no sine waves involved, only two heavily filtered impulses. The "error" in measuring the time arrival of either depends strictly on the "fuzz" in the plot, which is down some 150dB from peak. You can calculate time response to pretty much any sane level, with any of the three. Yes, the top line will be the "worst", in that it's about 10dB less energy.

Note that the input impulse (a 1 at the center point) is never restored, because I've removed all frequencies above 20kHz.

This is how you demonstrate and calculate time accuracy. Sine waves are only involved as basis vectors inside the mathematics.

With this illustration, I give up. If you won't actually go learn, please stop spamming with confused, circular arguments.

 

[bennybbbx]

I'll try this ONCE.

Here I have a plot of 3 impulse responses.

The first is 30-3000 Hz sampled at 96000
The second is 3000-20000 sampled at 96000 There is no overlap in frequency between the two plots.
The third is their sum, which in fact is an impulse of 30-20000

Now, this is a symmetric response, which you can come very close to getting out of a loudspeaker if you work hard and use the proper digital crossover, but in fact you'll see the same kind of result if you did this with a minimum phase filter (i.e. high level crossover with r's and c's and l's) tht was done correctly, except that the peak would come first. The summation would still result in similar results.

Note, there are no sine waves involved, only two heavily filtered impulses. The "error" in measuring the time arrival of either depends strictly on the "fuzz" in the plot, which is down some 150dB from peak. You can calculate time response to pretty much any sane level, with any of the three. Yes, the top line will be the "worst", in that it's about 10dB less energy.

Note that the input impulse (a 1 at the center point) is never restored, because I've removed all frequencies above 20kHz.

This is how you demonstrate and calculate time accuracy. Sine waves are only involved as basis vectors inside the mathematics.
View attachment 141147
With this illustration, I give up. If you won't actually go learn, please stop spamming with confused, circular arguments.

maybe you can answer this question with yes or no. Do you want say with this example that a slow bass mid speaker with soft suspension that reach 6 khz(0.16 ms period time is able to produce 2-3 khz waveforms with a time delay to original much less than 0.1 ms which is need for ITD regardless if the membrane need move much or less ?

then whats the reason wy stereo width of speakers is diffrent ?. i notice on headphone the step response look fastest. and the speaker with a fast looking step response do better stereo width i hear. here can hear example wav https://www.audiosciencereview.com/...of-the-synth-sound-in-the-attached-mp3.18401/

what i mean in real world a 6 inch or 8 inch speaker have much mass. this mass need move. this give delay much or less depend on how much it need move and how strong the magnets are and if hard or soft suspension. bass /mid woofers have to reach low bass frequency often soft suspension. because of the delay rise and fall time every speaker have, the membrane can not reach much higher frequency as 6 khz(0.16 ms) on slow speakers and do also lots of distortion when come near this frequency. when there is no ITD hear, then it doesnt matter if the 2-3 khz waveform do tear or call it maybe jitter about 0.1 or more ms depend on way it need move. but because ITD create out of the diffrence of the peaks between left and right speaker stereo hearing there are less delay and jitter need.

here for example i have a wave screenshot of frequency 50 hz with close bassport from kali. there can clear see it can produce 50 hz but, the waveform is not a exact sine and the peaks are wrong in compare to the input signal. this is of course extrem, but it can produce the frequency. and same happen more or less on mid frequency too. this can see when overlay original wave and recordet wave if it is more as 0.1 ms (4 samples). maybe should do such tests with triangle wav and can see better if speaker produce good lines

 

[j_j]

maybe you can answer this question with yes or no. Do you want say with this example that a slow bass mid speaker with soft suspension that reach 6 khz(0.16 ms period time is able to produce 2-3 khz waveforms with a time delay to original much less than 0.1 ms which is need for ITD regardless if the membrane need move much or less ?

Please give up this "period time" thing. It is not involved in this issue. Uniform delay in both channels of a system does not matter, either. Again, you do not understand basic Fourier mathematics.

 

[bennybbbx]

Please give up this "period time" thing. It is not involved in this issue. Uniform delay in both channels of a system does not matter, either. Again, you do not understand basic Fourier mathematics.

do you want say, a speaker have not more delay when it need move more way ?. do a closer look at wavescreenshot from brothers in arms at the line.

here can see that in real world the stereo channels are not uniform. in this case the right speaker need do more wayper sample as the right speaker. are you sure that this move is not delay more than 0.1 ms ?. if the speaker not delay this move more than 0.1 ms, wy this speaker can then not play 10 khz good ?. 0.1 ms is 10 khz. so a speaker that can reach all positions in less than 0.1 ms should be able to play 10 khz. for itd can also hear good 0.05 ms diffrence.


do you confirm that a speaker can reach higher frequency when the level is few ?.


this is the frequency response of a bass mid speaker 6.5 inch. it have a neodyn magnet and can be strong. http://www.loudspeakerdatabase.com/BC/6MBX44

did you really think this speaker do not add random delays depend on movement in range of 0.05 ms- 0.1 ms ?. you can see the overshoot region at 4.5 khz

 

[bennybbbx]

here is a good edxplain hgow can measure better the important parameter for speed of a speaker. so can really see if this Bass mid range speaker is precise enough to create mid range with better precision as 0.1 ms.

https://www.sweetwater.com/insync/square-wave-response/

the important thing for ITD is the slew rate. a speaker with slow slew rate create also on low frequency much diffrent timeshifts depend on the changes in signal amplitude per sample

A number of quantitative measurements can also be obtained from the square-wave response. Some of the most common measurements are:


Rise time: The time required for the signal amplitude to change from 10 to 90 percent of the total square-wave amplitude.


Overshoot: The amount by which the peak of the square wave exceeds its normal positive or negative amplitude. Large overshoots indicate peaking or excessive phase shifts in the high-frequency response of the device under test. Overshoot normally occurs only on the leading edges of a square wave.


Ringing: The tendency of band-limited square waves to oscillate on the peaks. This is largely a subjective measurement with the results stated as the severity of ringing. The greater the high-frequency peaking, the greater the ringing.


Tilt: A measure of low-frequency behavior. As low frequencies are filtered, phase shifts are introduced which cause the leading edge of the square wave to rise and the trailing edge to fall at more of an angle. This produces a tilt to the top and bottom of the square wave. The tilt is usually expressed as a percentage of the peak amplitude of the square wave.


Slew Rate: A measure of how fast a signal changes from one instantaneous value to another. The ideal square wave changes from one amplitude extreme to the other instantly. Practical devices cannot keep up with this transition and will often have a maximum speed with which they can change. This speed limitation will result in a tilted straight-line portion of the square wave edge.

 

[audio2design]

Benny, I have a EE Master's, and an M Sci. in electroacoustics. I was well on my way to my PhD, primarily doing work broadly grouped (for lay people) as ambisonics, but dabbled in some neuroacoustics mainly as my background made me the go to person for aspects of experimental design, and got sidetracked into some wok on preference correlation in audio. The only reason I didn't finish my PhD was a great opportunity to build a company. j_j's resume, especially on published research and academia as it relates, is I am sure many pages longer than mine, and his current "sideline" if I am interpreting properly is encoding for multi-channel audio delivery in 3D space (sorry if I simplified). In other words, we are not guessing.

Slew rate is not the most important thing for ITD. The most important thing from a single speaker standpoint is THD and IMD of the drivers contributing to the range of ~ approximately 200-1500Hz coupled with no rapid changes in the phase (over frequency) in this area. Due to envelope modulation of higher frequencies, there is an argument for other frequencies, but you have less spectral content in music and with odd exceptions, much lower distortion as well in the drivers. However, it is not a single speaker that is the most important. We are working with two channel audio, and even though most music is very lacking in ITD information, and there are limitation inherent from traditional capture/playback, the most critical thing, as what are "hearing" is the timing delay between the arrival of the left and right speakers (for ITD), is consistency between the two speakers w.r.t. phase over the critical frequency range listed above assuming the single speaker issues are taken care of.

 

[bennybbbx]

We are working with two channel audio, and even though most music is very lacking in ITD information, and there are limitation inherent from traditional capture/playback, the most critical thing, as what are "hearing" is the timing delay between the arrival of the left and right speakers (for ITD), is consistency between the two speakers w.r.t. phase over the critical frequency range listed above assuming the single speaker issues are taken care of.

if you think most music lack ITD information. then this music contain no reverb. but most music is record with reverb. this create the room and the ITD information for lower frequencies. read about how can create reverb. it is easy. just small delays of diffrent ms on left and right channel and level. this create the ITD for low frequency. so it sound on fast speakers that you really sit in the room(for example dire straight brothers in arms) . frequency from 200 hz upto 1000 hz are very important to create a good sounding room information. try highpass in music and hear how it sound. worse room then i hear.

you have written in earlier posts

Of course I can hear ITD, everyone can heard ITD (to my knowledge). Well hear is not accurate, I perceive location based on it is more accurate.

you mean that everyone can hear ITD ?. Does this mean that you have not hear the ITD tests i link if you really hear ITD ?. i dont think that everyone can hear ITD, because when can hear position diffrence of 0.05 ms, this is 20 khz frequency. simular to 3D view. there are many that see no 3D. so 3D television is not so much success.
ears need process small periods or high frequency. it is know that people who hear not good first have problems to locate position of speech for example. this is called cocktail party effect. search in inet about this. also there is known that yound people have often damaged ears. so it is good to do a test.

its also possible that i have a little loss in hearing but small bass mid speakers that are test as wide stereo and depth of field are fast enough so i get optimal stereo width. and the LP6 is worse for me. it sound as a reverb canceler. if somebody think the room in headphone sound wider as in speaker, then a fast speaker can help. this i hear clear.

https://www.hear-it.org/more-young-people-hearing-problems

One in four has hearing loss One in four 18-44-year-olds using headphones reports hearing problems, a study shows. According to an American study, one in four young New York adults aged 18-44 reports hearing loss and hearing problems are found in 23% of people who use headphones at a high volume at least five days a week for four hours a day. Researchers conclude that if young adults frequently use headphones at a loud volume, they should be prepared to deal with ringing in their ears or hearing loss as a result.

I have not written what you write

Slew rate is not the most important thing for ITD

. I have written "the important thing for ITD is the slew rate. " this is mean in the link .(where rise time, overshoot .is talk about) but if a speaker have slow slew rate it have also higher THD. how much delay diffrence a speaker have in relation the way it move is not test ansd how much ms a distortion of 0.3% for example a peak can delay.
most here are anonym. you too. I do not post my sourceforge account in which can see that i develop reverb other audio effects and some more programs.

additional distortion is not test on many volume levels. then can see that distortion change. and because left and right speakers need always diffrent ways move at same time, the THD change of a speaker depend on level is also a problem. so need measure. a speaker that change from 50 db- 90 db the distortion not much is better for ITD

 

[MakeMineVinyl]

Please give up this "period time" thing. It is not involved in this issue. Uniform delay in both channels of a system does not matter, either. Again, you do not understand basic Fourier mathematics.

There you go using logic again.

 

[audio2design]

0.05msec for the last time is not 20Khz. It is 0.05msec! Period. It is a delay. If you can't wrap your head around that then you will never understand this and just wasting time.

Reverb in recordings is normally used to generate artificial wall reflections, not to generate ITD. That is what gives you the sense of room space. ITD give you left to right positions of, in music direct sounds. It is not the only mechanism.

Again there is so much wrong in what you wrote there is no point addressing it individually as we are back to basics. P.s. knowing the slew rate gives you absolutely no information about THD

 

[bennybbbx]

0.05msec for the last time is not 20Khz. It is 0.05msec! Period. It is a delay. If you can't wrap your head around that then you will never understand this and just wasting time.

Reverb in recordings is normally used to generate artificial wall reflections, not to generate ITD. That is what gives you the sense of room space. ITD give you left to right positions of, in music direct sounds. It is not the only mechanism.

Again there is so much wrong in what you wrote there is no point addressing it individually as we are back to basics. P.s. knowing the slew rate gives you absolutely no information about THD

I have post you the link of a online converter that convert period time in frequency . please look at this link . so it is possible to convert period to frequency. frequency is a time measure and period time or ms too. it is with ITD same as with pictures. there happen sometimes not high cycling frequency for example but the transients need exact or it give tearing

when the artifical wall reflections are create and a transient in the midlle happen then depend on distance to wall. then it happen that the transient sample is for left channel delay 20 samples. the transient sample(which is in middle) for right channel is delay 25 samples. then there is ITD information in and it create a room feeling.have you a DAW install to test ?. which DAW you have ?. you can also use a simple delay that have a knob for feedback and allow the delay times set diffrent for right and left channel. when set diffrent delay values for left and right channel you get room feeling.

here is wide roomsound with delay and panner, maybe you have hear that some music sound as it come more right as the speaker ist. such an effekt is also ITD because when left speaker produce no sound with opposite phase you always hear the sound come from right speaker when left speaker is silent. see ambisonic. ambisonic can allow stereo speakers place very near but create a big room. maybe have no success because many people not hear