Chuck H 11/13/2017 9:40 AM
Phase at frequency w/speaker emulator circuit (Juan?)
I'm working on a passive circuit to emulate speaker frequency response. I'm pretty happy with the EQ but I'm concerned about the phase differential between LF and HF (almost 360*) and was wondering what sort of problems it might cause in actual listening perception or electronically at the input of a mixer/PA. I know that phase error is basically a time lag and if I interpret what I've read correctly I'm dealing with about a 1ms differential. I don't think that's going to be terribly audible, but I don't have much experience with this. Below is a graph of the frequency response of a G12H with the plot for my circuit overlaid on top. Phase is indicated by the dash. It's pretty rough because of the overlay, but I hope the info gets across.

TIA

EDIT: Also, does anyone know the real world phase error for an average speaker?
 
Mike Sulzer 11/13/2017 1:11 PM
If the phase of the speaker response is similar, then I think there is no problem. Do you know the past of the speaker response? There might be no problem anyway; phase often does not matter, but it can.

EDIt: Not sure what you mean by phase error of a speaker, but speakers have phase shifts just as other electronic components do. In fact the phase response of a speaker can be complicated. I think Eminence has phase plots for many of its speakers.
 
Chuck H 11/13/2017 1:56 PM
Thank you Mike. By "error" I just meant any differential from ideal. I mean, you wouldn't want your LF appearing 300ms behind the HF. That's extreme and ridiculous of course. Just illustrating. I know Eminence has impedance plots. I'll check there and see if they have phase plots as well. This circuit is strictly for EQ. It won't be a load for anything and it won't be driving anything with current. I did manage to find some speaker phase plots for high end audio stuff and it looks like phase relative to frequency can shift up to 150*. I'll bet a guitar speaker, with it's harder roll off top and bottom is worse. I only wanted to know if there was some inherent problem with phase shifts approaching 180* with a HF/LF differential approaching 360* for EQ purposes.
 
J M Fahey 11/13/2017 3:20 PM
What Mike Sulzer said: speakers have terrible phase shift problems, in any case your electronic circuit will always be better than the mechanical version, worst case will approach it, so thatīs what we are used to hearing anyway.
 
Chuck H 11/13/2017 3:27 PM
Thanks guys.
 
glebert 11/13/2017 3:38 PM
Dumb thought: presumably at some point after this emulation circuit someone is going to actually listen to the signal (either directly or from a recording), and they will be doing that through a speaker. If you try to build in the absolute phase response of a particular speaker once it goes through the final speaker you may end up with "extra" phase impact.
 
catalin gramada 11/13/2017 3:51 PM
Quote Originally Posted by J M Fahey View Post
What Mike Sulzer said: speakers have terrible phase shift problems, in any case your electronic circuit will always be better than the mechanical version, worst case will approach it, so thatīs what we are used to hearing anyway.
Yup.No problems normal btw phasing problems could be when sound reinforcement request special for low freq.Slaving works,micing could be dificult and needs phase adjustments
 
R.G. 11/14/2017 6:43 AM
There is a hidden truth lurking under this thread. That is, there can be no filtering without phase shift, at least in analog electronics. The fundamental way that filtering happens is with the interaction of resistive (i.e. no phase shift) elements and reactive elements, things which have a differing impedance with frequency, and that effect by its very nature causes phase shift. The same thing happens in mechanical, acoustic, etc. systems.

It is probably possible to use DSP programming to affect amplitude as a function of frequency and then to correct signal phase back to no phase shift (or any arbitrary phase shift) but the normal sorts of digital filters also introduce phase shift with amplitude variations.

So if you want filtering, you get phase shift too.
 
J M Fahey 11/14/2017 7:10 AM
Quote Originally Posted by glebert View Post
Dumb thought: presumably at some point after this emulation circuit someone is going to actually listen to the signal (either directly or from a recording), and they will be doing that through a speaker. If you try to build in the absolute phase response of a particular speaker once it goes through the final speaker you may end up with "extra" phase impact.
True.
That said, using a speaker emulator , which to be more precise should be called a "guitar" speaker emulator , or it would not be needed to begin with, sort of implies that final sound will be played through a Hi Fi, Studio or, worst case, PA speaker ... all of which have (or try hard to) flattest response and minimal phase shifts.
And we add the speaker emulator to that (flat but unexciting) mix precisely to add that off taste flavour we like
 
Mike Sulzer 11/14/2017 12:32 PM
Quote Originally Posted by R.G. View Post
There is a hidden truth lurking under this thread. That is, there can be no filtering without phase shift, at least in analog electronics. The fundamental way that filtering happens is with the interaction of resistive (i.e. no phase shift) elements and reactive elements, things which have a differing impedance with frequency, and that effect by its very nature causes phase shift. The same thing happens in mechanical, acoustic, etc. systems.

It is probably possible to use DSP programming to affect amplitude as a function of frequency and then to correct signal phase back to no phase shift (or any arbitrary phase shift) but the normal sorts of digital filters also introduce phase shift with amplitude variations.

So if you want filtering, you get phase shift too.
Yes, there is no problem constructing a digital filter with no phase shifts, but you might not like the transient response! There are only so many free parameters, and if you specify the amplitude and phase as a function of frequency, do not expect anything else to be what you might want. You might think of such a filter as using the Fourier transform (in some clever way so that finite length transforms can coupled together to give a continuous signal), and you can modify the Fourier coefficients in amplitude and phase as you wish, and then transform back to the time domain. But as for the time domain response, you get what you get. Remember, phase is quite audible if introduced in a correlated way over a range of frequencies. The simplest example is time stretching, where you can take a short transient and make it much longer while keeping the high frequencies, and changing only the phase. The result sounds nothing like the original. On the other hand, taking a musical instrument signal and shifting the relative phase of the various harmonics of a note can produce very little effect if done right, and played through a linear system. The waveform shape is modified, of course, and so if here are gross nonlinearities (guitar amp played loud) then the harmonics added by that distortion are a function of the waveform shape to some extent. So this can get very complicated.