THE IMPORTANCE OF TIME DOMAIN RESPONSE MUST READ


Speaker designers ignore or downplay the importance of TIME. Why?

A high end speaker should be as accurate as possible and that means it should not only be optimized with regard to frequency response but time response.

Back in the 70’s and around that time, speaker engineers thought that a perfect speaker would be one that had a flat response. This idea has waxed and waned in popularity over the years and even now there is no consensus.

What the speaker engineers forgot to consider is Time response.

The time reponse of a speaker is how fast it starts and stops. A perfect speaker would have a perfect time response of 0. Since this is not possible, we must get as close to it as possible. The problem is speakers engineers have neglected this aspect of the design and so speakers over the last 40 years have not improved in this respect.

Time is such an important aspect of the sound we hear. We not only hear tone but also time. The brain can detect time differences of only a few microseconds. Experiments have shown that the start of each note is what we use to determine what instrument is producing that sound.

We must ensure that our crossovers do not smear the time response because it will be heard by our ears. Time inaccuracy is why high end speakers do not sound like real instruments.

Diffraction from the cabinet can also cause time smear. We need spherical cabinets not square boxes. Tweeters need to be time aligned in order to ensure that when the woofer stops so does the tweeter. When the woofer starts, so must the tweeter. The woofer itself has to have a Qts of ZERO to prevent time smear. Ports must not be used or else you will get ringing.

We need to make it mandatory for speaker companies to publish the time response of all their speakers so that consumers can easily compare and decide exactly what they want. Some may actually prefer a speaker that has a poorer time response and that is fine. The problem is, we cant decide unless we know what we are buying can we?

Unfortunately, 90% of speakers on the market, even high end speakers have ports. And they are also made of cheap wood, even though there must be better materials by now. Some materials ring more than others.

So dont be deceived folks. If you want better speakers, you will probably have to make them yourself because speaker manufacturers dont care about sound quality. They spend millions of dollars on anechoic chambers all so that they can get a flat response but they spend zero effort on better time domain response. We are being duped.

kenjit

Showing 8 responses by holmz

Of course they dont know what they’re doing. A speaker is supposed to reproduce the signal you give it. Yet all high end speakers sound and even measure differently. They can’t all be right can they?

@kenjit How wold you define “right”?
Are there some specific measurements that show when one is more right than another?

 

Maybe @roxy54 - or the air pressure response of a driver given an impulse of current or voltage…

But the later assumes that the current is somehow not allowed to ring in the circuit from the mass of the driver and its motor through the cable to the amplifier.

Hopefully @kenjit can clarify what he means, so we are not left guessing.

 

Then there is:

The time response of a speaker is how fast it starts and stops

So it that a speaker, as in cabinet, driver… i.e. the whole enchilada?
or “speaker” as in a single “speaker driver” unit?

(where “speaker driver unit” is like an indivisible quanta fundamental element (i.e. unit) of what makes up an elemental speaker.)

we need to look at the cumulative spectral decay to ensure that its 0ms at all frequencies from 20hz to 20khz. I've

What is oms?
(Is that 0 milliseconds?)

I do not think that we can apply an impulse to a system and get it to decay instantaneously unless we have zero output impedance, which a passive crossover  sort of gets in the way of.

The spectral decay is amplitude based in frequency versus time.
So yes it is a time domain based description, but it is shown in a PSD tyle of frequency response.
Is there a phase or complex component lurking around somewhere?

 

Lets assume that we have two speaker, both with 0 msec of decay.

Are there other measurements that would inform us of whether one is better than the other?
Say ones that included some idea of phase or time domain response (without frequency response), or do we just use frequency response? 

There are many sources of ringing within the speaker. The cabinet can ring, the driver can ring, the crossover parts can ring. We need to stop all ringing wherever it occurs.

Maybe let’s just ignore the ringing that may be happening long after the square wave’s fundamental period is over.

Or we could take a step function, so we only have 1/2 of the square wave… and that makes it simpler to consider and we got rid of half of the square wave… and also got rid of a train of square waves.

 

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It needs to be able to reproduce a square wave

What needs to reproduce a square wave?

  1. The position of cone or the velocity or the acceleration?
  2. or the air pressure

What should the electrical square wave input look like in a plot?
What are we supposed to be measuring… and how?

@kenjit earlier you were talking about:

There are many sources of ringing within the speaker. The cabinet can ring, the driver can ring, the crossover parts can ring. We need to stop all ringing wherever it occurs. It needs to be able to reproduce a square wave

Do you have the square wave measurements to show for your speaker?

Or is it a Gedanken design?

 

Now time domain should relate to a wide band response. If for rxemple you have a two ways, the two point of emissions should arrive at the ears at the same time, so they are in phase and time correct. But it’s important to consider the behavior of the cross-over (inductive and capacitive parts. 6db slopes allowing for a better time domain behavior (less phase rotations) than 12 or 18 db slopes.

The position of the tweeter and woofer have to be such that cross-over frequency and behavior are taken in account. Introducing a square wave and measuring it through a microphone (able to reproduce it) can help to finalize the position of the tweeter. The ear is very sensible to time domain by the way

@armagedon36 ^agree^
We could break it down into before and after…

  • With “before” being the launching of the acoustic field off of the transducer, which you are mentioning is best when it is faithful to the input signal (e.g. square wave).
  • and the “after” being any resonance or ringing after the signal should have ended.

But… the OP at one point was talking more about cabinet ringing after the fact, and having a death grip on the driver with aa QTS=0 and did not seem to care about the “before” part.

At least in chronological order, worrying about what happens later, is a bit
“after the fact”… as you implicitly pointed out.

It is probably best to start off right, and not have it go all down the sewer with cabinet ringing and resonances later. (IMO)

 

But we do not know what the OP was thinking as they went silent pretty quickly.

@erik_squires - one of the links mentions 1st order is “as good as it gets without going active filters and bi amping.”

Do you have opinions on:

  • Active as in DSP PEQs, FIR etc.
  • Active as in OP-Amp based crossovers?
  • other?