The attached show what I mean. I simulated a first cycle with a sine source switched on at T=0 with a voltage controlled switch.
The original first cycle is in green and shows the sudden start-up of the signal.
Next I simulated a 24dB xover (the opamps are just x1 buffers to isolate sections, output in red.
You see that the initial attack is much more gradual now, and also that the signal amplitude is less than the 'ideal' first cycle. So that is what the woofer sees, and that is what I believe the woofer has no problem with.
jd
The original first cycle is in green and shows the sudden start-up of the signal.
Next I simulated a 24dB xover (the opamps are just x1 buffers to isolate sections, output in red.
You see that the initial attack is much more gradual now, and also that the signal amplitude is less than the 'ideal' first cycle. So that is what the woofer sees, and that is what I believe the woofer has no problem with.
jd
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Hi Janneman,
The most realistic test signal is the bass guitar or drum itself !
Quite right! But what is shown as a 'first cycle' curve is much more extreme, unrealistically so, in my opinion. Please see graph in my post above.
jd
I also looked at what would go to the tweeter, by taking the difference between the original graph and the woofer part (blue).
You can clearly see that the 'fast' part (blue) is split off and will go to the tweeter, the lf part (red) goes to the woofer. Adding red and blue gives you the original again. (The top opamp is a unity gain difference block).
jd
You can clearly see that the 'fast' part (blue) is split off and will go to the tweeter, the lf part (red) goes to the woofer. Adding red and blue gives you the original again. (The top opamp is a unity gain difference block).
jd
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Hi Jannerman,
You are completely missing the point.
It is the driver response in its own right which is the problem, and this has nothing to do with crossovers.
The transduction distortion due to kinetic/ potential energy exchanges/ alternations during music time are independent of, and quite different to, the waveform separations you illustrate.
How then would you explain the T-bass improvement of reproduction when used with a fullrange driver, where all waveforms are simultaneously present at the driver terminals?
You are completely missing the point.
It is the driver response in its own right which is the problem, and this has nothing to do with crossovers.
The transduction distortion due to kinetic/ potential energy exchanges/ alternations during music time are independent of, and quite different to, the waveform separations you illustrate.
How then would you explain the T-bass improvement of reproduction when used with a fullrange driver, where all waveforms are simultaneously present at the driver terminals?
Hi Jannerman,
You are completely missing the point.
It is the driver response in its own right which is the problem, and this has nothing to do with crossovers.
The transduction distortion due to kinetic/ potential energy exchanges/ alternations during music time are independent of, and quite different to, the waveform separations you illustrate.
How then would you explain the T-bass improvement of reproduction when used with a fullrange driver, where all waveforms are simultaneously present at the driver terminals?
Hi Grahan (it's 'janneman' btw, without the 'r'),
Yes maybe I miss the point. I was under the impression that you were trying to correct something that was the result of a 'first cycle' distortion, however defined.
My point was that the first cycle you showed to illustrate the issue is not a realistic signal and therefor the 'problem' is only artificial and doesn't exist in real music with real speakers.
Are we miscommunicating? What is the relevance of 'first cycle' to what T-bass tries to do?
jd
I look at Linkwitz' site and what I found in this context is his cosine-shaped tone bursts. I agree that these are realistic signals; because of the gradual envelope build-up the signal is much more benign for a driver, and I doubt that the effect that T-bass tries to 'fix' is there at all with a shaped tone burst.
Is this the graph you referred to? If not, could you provide a link to what you meant?
jd
No, I don't think the reference was to the cosine shaped bursts, but to 4 cycles bursts here in the context of the Linkwitz transform circuit:
Active Filters
(scroll down a bit)
There you can see the efect of the first half cycle in a 4-cycles burst having a lower amplitude.
No, I don't think the reference was to the cosine shaped bursts, but to 4 cycles bursts here in the context of the Linkwitz transform circuit:
Active Filters
(scroll down a bit)
There you can see the efect of the first half cycle in a 4-cycles burst having a lower amplitude.
Are you referring to the section 9 titled 'Linkwitz Transform (biquad)'?
jd
You can clearly see that the 'fast' part (blue) is split off and will go to the tweeter, the lf part (red) goes to the woofer.
It will be more obvious, if you use a signal that consists of at least two different frequencies. In your example the tweeter sees the same signal shifted in phase, but with a higher amplitude than the woofer, because the crossover frequency is below the signal's frequency. There is no 'faster' and 'slower' part in a sine signal with a constant frequency.
[snip]There is no 'faster' and 'slower' part in a sine signal with a constant frequency.
Of course, but when you use a tone burst, there are several harmonic components that are routed to the appropriate driver depending on freq and xover, that was what I wanted to illustrate.
So it is not appropriate to try to equalize a woofer to handle accurately and completely the tone burst, because in real world it will never see the tone burst, it will see the lf components in the tone burst.
jd
you forget about the "envelope"... a "tone burst" isn't a single tone, even if its non-zero part is a sine wave. The tone burst is actually a sine wave amplitude-modulated by a square wave, thus it has an (theoretically) infinite spectrum in the frequency domain.
Indeed. And a low-pass filtered square wave at that, in the context of an audio band.
jd
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If we pick a bass guitar string, it does not leave the plectrum with infinite speed.
The harmonics might be correctly reproduced by mid/ tweeter, but the LF OB driver(esp) starts from zero momentum and then resonates/increases in amplitude within the first few 1/2 cycles.
Indeed, a picked string does not accelerate from rest to maximum velocity instantly. In that case, there is no "first half cycle" effect to correct. The LF driver will (more or less) accurately reproduce the string movement as picked up by the pick-up. Which, incidentally, can only sample part of the string's full suite of resonances. When we get to acoustic bass, the sound-box excitation (and thus, most of the sound) is generated from the end of the string which has a different balance of harmonics.
Some light bedtime reading:
TIMEDOMAIN - Theory & Technology
(Actually, you ought to make sure you're well rested first. English is not the author's natural language, and he appears to be working with an incomplete understanding of the underlying physics. It's still food for thought, though.)
This section has particular relevance to the current discussion:
TimeDomain - Theory & Technology
This is why bass can sound much more realistic through good headphones - tiny mass/ momentum, whilst putting your head closer to an uncompensated LF driver still does not make LF reproduction sound any more realistic !
You have to consider that a headphone driver is usually operating above its fundamental resonance. Its acoustic pressure phase at LF will be opposite to that of an OB driver operating at or below resonance.
But more importantly, it is a full-range driver. It is reproducing much more of the spectrum than a LF driver does. To better compare the two, you need to LP filter the signal to the headphones to mimic the frequency response of the LF driver plus crossover.
Plain amplitude compensation cannot compensate for what the driver does either because the driver has already distorted the waveform cycle in real-time. It takes an inverse distortion in real time to counter the driver error. Once the driver is compensated, then further amplitude compensation becomes possible, as long as phase distortion does not then result.
Something like this?
http://www.diyaudio.com/forums/multi-way/121385-loudspeakers-room-system-88.html#post2127720
It shows how inverse equalisation can compensate to a large extent for distortion of the waveform caused by energy storage in a resonance. It "accelerates" the driver at the beginning of the signal and, just as importantly, applies an inverse effect ("braking") at the end. When I get a round tuit, it'll be interesting to see if T-bass does the same thing. If it affects the beginning and end of a tone burst as in the example quoted above, then it's an inverse equalising filter. If it only affects the beginning of the burst and not the end, then it's something else.
Hi Don, I followed your last link.
I do not know what anyone there is doing or illustrating.
Possibly so much theoretical waffle !!! but so little real-world understanding.
Thus I have no wish to become involved.
Room compensation is not possible -
only a palliative attempt to compensate for ONE listening position in a room, which then ruins reproduction throughout the rest of the room, because compensation applied to correct driver plus room response for one listening position, will by necessity already make that same driver radiate erroneously for propagation paths to differently positioned listeners.
No matter what a room does, we cerebrally discern leading edge waveform dynamics separately from room influenced secondary responses. Modifying radiated dynamics to counter propagated tonal effects of and within a roomat any single point immediately becomes seperately discernible as a distortion to the dynamic response of the source from other listening positions.
I do not know what anyone there is doing or illustrating.
Possibly so much theoretical waffle !!! but so little real-world understanding.
Thus I have no wish to become involved.
Room compensation is not possible -
only a palliative attempt to compensate for ONE listening position in a room, which then ruins reproduction throughout the rest of the room, because compensation applied to correct driver plus room response for one listening position, will by necessity already make that same driver radiate erroneously for propagation paths to differently positioned listeners.
No matter what a room does, we cerebrally discern leading edge waveform dynamics separately from room influenced secondary responses. Modifying radiated dynamics to counter propagated tonal effects of and within a roomat any single point immediately becomes seperately discernible as a distortion to the dynamic response of the source from other listening positions.
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I fully agree with that. I hesitate to mention how many hours I tried to equalize my listening room, using one of those DSP auto/manual correction systems (DEQX PDC 2.6), until it finally dawned on me.
If you have say a 'dip' in the freq response at a particular freq, at the listening position, caused by room reflections, you can equalize it with a corresponding 'pip' at that freq.
But, if you go to a different listening position, the dip occurs at a different freq because of different time delays in the reflections. So, what you end up with is the 'pip' of the correction where it is not needed, and another 'dip' elsewhere that is not corrected. You're working backwards big time!
jd
If you have say a 'dip' in the freq response at a particular freq, at the listening position, caused by room reflections, you can equalize it with a corresponding 'pip' at that freq.
But, if you go to a different listening position, the dip occurs at a different freq because of different time delays in the reflections. So, what you end up with is the 'pip' of the correction where it is not needed, and another 'dip' elsewhere that is not corrected. You're working backwards big time!
jd
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Hi Don, I followed your last link.
I do not know what anyone there is doing or illustrating.
The illustration at the link shows what happens when a tone burst is applied to a resonant system (speaker driver, or room). It then shows what happens when you apply a filter with an inverse response - for example, if the resonance has a 10 dB peak and a Q of 0.8, applying a notch filter with a notch of 10 dB and a Q of 0.8 will largely cancel the resonance. It compensates both for the energy storage at the beginning and the release at the end. Do you think that is very similar to what a T-bass does?
Actually, your T-bass descriptions focus on what happens at the beginning of the signal. You say the T-bass adds energy during the first half cycle or so to compensate for the energy storage in the driver that prevents it reaching "full amplitude" in the first half cycle. What happens to that added energy? When is it released?
Possibly so much theoretical waffle !!! but so little real-world understanding.
Thus I have no wish to become involved.
We all have a "real world understanding" of what time the sun will rise tomorrow morning, but few of us can explain it in orbital mechanics terms - the equations which predict the observed effect. Without the equations, we're working purely from observed behaviour - in other words, tweak trhe values until we get the desired result. With the equations, we can solve the problem for any situation - for example, what time the sun will rise on New Years Day 2015.
The point is that an understanding of the science behind the T-bass operation will allow deriving equations and models which can predict the component values required to obtain optimum T-bass performance for any given driver and baffle combination.
Room compensation is not possible - (...)
My mistake - I should have made it clearer that the reference was only to the graph showing what happens when a sine burst is applied to a resonant system, which is relevant to this discussion.
Regards,
Don.
Don,
Ever heard the expression - 'The Lord helps those who help themselves' ?
Will you PLEASE try this circuit *hands-on* because your ears will instruct your brain far better than your present interpretation of theory as based upon circuits and methods you already understand . !
Now you ask >> What happens to that added energy? When is it released? <<
So I ask in return - **What added energy ?** and I reply -
The transformer steps up the leading edge due to the step-up ratio of the transformer, this via the choke on the grounded end.
The series C+L then resonate and lift the impedance of the grounding reference thus reducing step-up after the first 90 degrees.
The resistors and winding losses damp the circuit (Q) as necessary and dissipate energy so that it does not (cannot) remain dynamically active.
I am hoping that today's Sun where you live will indeed provide a fresh dawning.
(My notes have already covered the C+L as being a series tuned shunt, but the thread is now so full of criticism, often personally aimed, by those who say it can't work, like you Don, so I don't know how anyone can any longer easily find information they need, and I have so much else to cope with I now see this thread as being ruined.)
Ever heard the expression - 'The Lord helps those who help themselves' ?
Will you PLEASE try this circuit *hands-on* because your ears will instruct your brain far better than your present interpretation of theory as based upon circuits and methods you already understand . !
Now you ask >> What happens to that added energy? When is it released? <<
So I ask in return - **What added energy ?** and I reply -
The transformer steps up the leading edge due to the step-up ratio of the transformer, this via the choke on the grounded end.
The series C+L then resonate and lift the impedance of the grounding reference thus reducing step-up after the first 90 degrees.
The resistors and winding losses damp the circuit (Q) as necessary and dissipate energy so that it does not (cannot) remain dynamically active.
I am hoping that today's Sun where you live will indeed provide a fresh dawning.
(My notes have already covered the C+L as being a series tuned shunt, but the thread is now so full of criticism, often personally aimed, by those who say it can't work, like you Don, so I don't know how anyone can any longer easily find information they need, and I have so much else to cope with I now see this thread as being ruined.)
Hi there
Having built this circuit, i can confirm that it actually works. I have used this to augment bass in the Open baffle using Eminence Beta 15. Whether a difference is pronounced or subtle, depends on the rest of your setup (amp and the speakes). You need a powerful SS amp in order to make this work, otherwise you won't notice the effect as it is supposed to do.
To make the equivalence of this, imagine that you put wider tyres to your car to provide a better friction and smoother ride. But they will present a load to the engine, and if you have an underrated engine, the effect can be quite the opposite. For non-believers, please take the whole picture into account before saying that it doesn't work. I will not involve myself into any arguments or theory, I have no time for that.
Graham, I can just express my satisfaction with the circuit and thank you for sharing it with us.
Regards,
Vix
Having built this circuit, i can confirm that it actually works. I have used this to augment bass in the Open baffle using Eminence Beta 15. Whether a difference is pronounced or subtle, depends on the rest of your setup (amp and the speakes). You need a powerful SS amp in order to make this work, otherwise you won't notice the effect as it is supposed to do.
To make the equivalence of this, imagine that you put wider tyres to your car to provide a better friction and smoother ride. But they will present a load to the engine, and if you have an underrated engine, the effect can be quite the opposite. For non-believers, please take the whole picture into account before saying that it doesn't work. I will not involve myself into any arguments or theory, I have no time for that.
Graham, I can just express my satisfaction with the circuit and thank you for sharing it with us.
Regards,
Vix
Hi there
Having built this circuit, i can confirm that it actually works.[snip]Vix
I don't think that was an issue at all. Of course it works, how could it not?
jd
Thank You Vix.
As you say it works best with a good SS amplifier, and better still with loudspeaker system sited SS monoblocks.
I also think it likely the Beta 15 would provide so much more accurate (tighter and better controlled) LF reproduction than the Alpha 15 on OB, especially if T-bass driven.
As you say it works best with a good SS amplifier, and better still with loudspeaker system sited SS monoblocks.
I also think it likely the Beta 15 would provide so much more accurate (tighter and better controlled) LF reproduction than the Alpha 15 on OB, especially if T-bass driven.
- Home
- Loudspeakers
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- 'T'-bass drive for OB LF drivers.