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Thread: Phase at frequency w/speaker emulator circuit (Juan?)

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    Supporting Member Chuck H's Avatar
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    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?
    Attached Thumbnails Attached Thumbnails graphcomp1.gif  
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    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.
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    Supporting Member Chuck H's Avatar
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    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.
    Last edited by Chuck H; 11-13-2017 at 10:06 PM.
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    Old Timer J M Fahey's Avatar
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    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 that´s what we are used to hearing anyway.
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    Juan Manuel Fahey

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    Supporting Member Chuck H's Avatar
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    Thanks guys.
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    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.

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    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 that´s 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
    Last edited by catalin gramada; 11-14-2017 at 12:03 AM.
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    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.
    Amazing!! Who would ever have guessed that someone who villified the evil rich people would begin happily accepting their millions in speaking fees!

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    Old Timer J M Fahey's Avatar
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    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
    Juan Manuel Fahey

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    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.
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    Supporting Member Chuck H's Avatar
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    Fantastic and thank you to all for continuing discussion on the topic. FWIW I absolutely do intend to do listening tests and the final circuit will allow switching the emulator circuit out for use with either sound reinforcement or a guitar speaker cabinet. My assumption being that sound reinforcement speakers are designed for a flat(ish) response. I could well find this to be false to a greater or lesser degree requiring circuit modification.
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    Quote Originally Posted by Chuck H View Post
    ...I'm concerned about the phase differential between LF and HF (almost 360*)
    If you look at the low frequency response of a typical (sealed enclosure or free air) speaker, it has the form of a second-order high pass filter. In a perfect world, that would be accompanied by a total of 180 degrees of phase shift as you go through the resonance frequency of the speaker. There will be 180 degrees of phase lead well below resonance, and zero degrees phase shift well above resonance.

    I happen to be working on creating my own speaker simulation software at the moment, so I'll attach a screenshot showing this, for a fictional speaker with a resonance at 100 Hz.

    If you had an ideal infinitely stiff speaker cone, that would be the whole story. In practice, as you go higher in frequency, eventually you get cone break-up modes, which are themselves mechanical resonances, each one accompanied by 180 degrees of phase *lag* as you sweep through it (from a frequency well below, to a frequency well above).

    So if you were dealing with a nearly ideal speaker, with only its fundamental (bass) resonance, plus one single cone break up mode, you would already have 360 degrees of phase shift within the frequency spectrum - 180 degrees lead at very low frequencies, zero phase in the midrange, and 180 degrees lag at high frequencies well above that cone breakup frequency.

    Real life is far worse, with additional breakup modes coming thick and fast as you go up in frequency...each one accompanied by yet another 180 degrees in phase.

    And all this is if you were placing your ear right on top of the dust-cap. If you are at a normal distance from the speaker, there is additional phase shift as the sound travels through the air to your ears - three hundred and sixty degrees of phase shift for every wavelength travelled. The speed of sound in a home at normal temperature is around 340 metres/second, so if you were listening to a 3.4 kHz tone, and your ear was one metre away from the speaker, there would be ten wavelengths of sound between the speaker and your ear. This means three thousand, six hundred additional degrees of phase shift, on top of the 360+ in the speaker driver itself!

    Note, by the way, that if you were listening to 34 Hz (from your 5-string bass guitar, say), you are only one-tenth of a wavelength away, so only 36 degrees of phase. In other words, very little phase shift at 34 Hz, but lots and lots of phase shift at 3.4 kHz...

    In other words, at one metre distance from the speaker, there is more than three thousand degrees of phase shift between deep bass and mid-treble, just because of the way sound behaves when it travels through air!

    (And, at a more realistic listening distance, there may be two or three or four times as much - literally, over ten thousand degrees of phase shift between bass and treble, even if you were in an anechoic chamber!)

    All this is why I pay no attention to most of the Audiophool fussing over speaker phasing. Phase only really seems to matter when you have two or more drivers simultaneously emitting the same signal, with a phase shift between them. In that particular case, the multiple signals will interfere with each other, and cause peaks and dips in the frequency response.

    This sort of thing happens during the crossover region in Hi-Fi speakers (where both woofer and tweeter are emitting the same sound), and it happens all over the spectrum if you stuff four Celestions in one cab and drive them all with a full-range guitar signal.

    But one speaker (or one simulated speaker) by itself? Your ear doesn't care. The big bass drum in the marching band sounds the same whichever side of the road you happen to be standing on when the parade goes by. Clear proof that 180 degrees of phase-shift in the bass makes no difference whatsoever to the way it sounds!

    And the million dollar question: this speaker emulator is for a project involving running a micro valve guitar amp direct into a P.A. system, perhaps? Any nifty stuff to share?

    -Gnobuddy
    Attached Thumbnails Attached Thumbnails spkr_amplitude_n_phase_response.png  
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    Quote Originally Posted by Gnobuddy View Post

    Note, by the way, that if you were listening to 34 Hz (from your 5-string bass guitar, say), you are only one-tenth of a wavelength away, so only 36 degrees of phase. In other words, very little phase shift at 34 Hz, but lots and lots of phase shift at 3.4 kHz...

    In other words, at one metre distance from the speaker, there is more than three thousand degrees of phase shift between deep bass and mid-treble, just because of the way sound behaves when it travels through air!


    -Gnobuddy
    Maybe I'm missing something, but I don't think that is the way it works. Propogation delay is not the same as phase shift.

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    Quote Originally Posted by glebert View Post
    Maybe I'm missing something, but I don't think that is the way it works. Propogation delay is not the same as phase shift.
    For a wave moving in time (sound wave, in this case), time delay and phase delay are inextricably linked. If one cycle lasts for, say, one millisecond, then one millisecond causes 360 degrees of phase increase.

    Put another way, there is relative phase shift (between, say, two different sine waves at the same frequency). There is also absolute phase - the "wt" in the equation Y = A sin(wt). It takes 360 degrees of phase to make one one full wave, i.e, wt increases by 360 degrees from its initial value to create one full wave. There will be 360 more degrees of phase for each subsequent wave.

    Still another way to think of it: put one microphone a half-wavelength further from the source than another, and we agree that the two mics will put out signals 180 degrees apart, yes?

    Now move the further microphone another quarter-wavelength away, and now there will be 270 degrees phase shift between the two signals, yes?

    Keep moving the further microphone in little steps, say one-hundredth of a wavelength further each time. You get another 3.6 degrees of phase with each additional increment in distance.

    So what happens when the second microphone is a full wavelength further than the first? We kept increasing the phase shift beyond 270 degrees in steps. Clearly, the two signals are now 360 degrees apart in phase.

    360 degrees might look like zero degrees on an oscilloscope, but that's because oscilloscopes are not good tools for looking at total phase. Take Fourier transforms of those two time-delayed waveforms, and you will see the additional 360 degrees of phase in one of them.

    -Gnobuddy
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    If all the wave frequencies move at the same speed, then the shape of the total waveform does not change, and so the relative phases remain the same. This is what counts, and it is not the same as introducing frequency dependent phase shifts at the source.

    For example (https://brilliant.org/wiki/amplitude...r-phase-shift/), propagation of a simple wave can be described by

    sin(k(x - vt)) where:

    the phase is the argument of the sine,
    x is the spatial coordinate,
    t is the time coordinate,
    and k = 2*pi/(wavelength),
    v is the phase velocity.

    If the phase velocity is the same for all frequencies, the phase us the same at all frequencies for each x. (Frequency is not in the equation!)
    Then we have v = omega/k ,
    where omega is 2*pi*f,
    and the equation can be written:
    sin(kx - omega*t)

    This form contains the frequency explicitly.

    If v is a function of frequency, then the relative phase does change as a function of frequency.

    Quote Originally Posted by Gnobuddy View Post
    For a wave moving in time (sound wave, in this case), time delay and phase delay are inextricably linked. If one cycle lasts for, say, one millisecond, then one millisecond causes 360 degrees of phase increase.

    Put another way, there is relative phase shift (between, say, two different sine waves at the same frequency). There is also absolute phase - the "wt" in the equation Y = A sin(wt). It takes 360 degrees of phase to make one one full wave, i.e, wt increases by 360 degrees from its initial value to create one full wave. There will be 360 more degrees of phase for each subsequent wave.

    Still another way to think of it: put one microphone a half-wavelength further from the source than another, and we agree that the two mics will put out signals 180 degrees apart, yes?

    Now move the further microphone another quarter-wavelength away, and now there will be 270 degrees phase shift between the two signals, yes?

    Keep moving the further microphone in little steps, say one-hundredth of a wavelength further each time. You get another 3.6 degrees of phase with each additional increment in distance.

    So what happens when the second microphone is a full wavelength further than the first? We kept increasing the phase shift beyond 270 degrees in steps. Clearly, the two signals are now 360 degrees apart in phase.

    360 degrees might look like zero degrees on an oscilloscope, but that's because oscilloscopes are not good tools for looking at total phase. Take Fourier transforms of those two time-delayed waveforms, and you will see the additional 360 degrees of phase in one of them.

    -Gnobuddy
    Last edited by Mike Sulzer; 12-01-2017 at 01:15 PM. Reason: typo
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    Supporting Member Chuck H's Avatar
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    Quote Originally Posted by Gnobuddy View Post
    And all this is if you were placing your ear right on top of the dust-cap. If you are at a normal distance from the speaker, there is additional phase shift as the sound travels through the air to your ears - three hundred and sixty degrees of phase shift for every wavelength travelled. The speed of sound in a home at normal temperature is around 340 metres/second, so if you were listening to a 3.4 kHz tone, and your ear was one metre away from the speaker, there would be ten wavelengths of sound between the speaker and your ear. This means three thousand, six hundred additional degrees of phase shift, on top of the 360+ in the speaker driver itself!
    Wow! I was eminently aware of this phenomenon (but didn't know such impressive figures). As is anyone that's ever played different venues with different cabinet designs. I figured from the beginning that a blunt recreation of a speakers measured frequency response may well require some tweaking.

    Quote Originally Posted by Gnobuddy View Post
    And the million dollar question: this speaker emulator is for a project involving running a micro valve guitar amp direct into a P.A. system, perhaps? Any nifty stuff to share?
    Indeed it is. A full amp with OT and power tubes distilled down to a preamp. It can be used as the 2W amp it is or played intio an inductive dummy load followed by a line level out, compensated for speaker EQ or not via a switch so it can be used for anything from direct recording, into a PA or as a standard guitar rig preamp into a clean power amp and guitar speaker cab. Such an amp can create everything from clean to high distortion tones relative to pot settings and OVER ALL amp response. The only drawback is that there can be no "channel switching" per se since the power amp is a shared part of the gain structure for all tonal textures. So I'm setting up to work with another guy on digital pots and presets as a solution.

    All I have right now is a pencil and graph paper sketch of the analog circuit. When I get that part of it drawn up on computer media I'll post it. The digital stuff is out of my wheelhouse, but if there's any continued interest I'll update as final designs come into focus.
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    Supporting Member Chuck H's Avatar
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    That was my perception of it too (minus the math ). But that would be relative to all things being perceived at different distances once phase relationships are settled acoustically, right? In other words, with an open back cabinet sitting some distance from the wall behind in a small room... I would think that phase @ frequencies would be different at various vantages within that room, right? Only from outside the room (for example) would the phase relationships remain analogous @ distance.?.
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    Yes, in a room you have many paths between source and listener, and so the phase is very complicated when you add them all up. If the human ear-brain did have high sensitivity to phase, it would be very distracting when you move.

    Quote Originally Posted by Chuck H View Post
    That was my perception of it too (minus the math ). But that would be relative to all things being perceived at different distances once phase relationships are settled acoustically, right? In other words, with an open back cabinet sitting some distance from the wall behind in a small room... I would think that phase @ frequencies would be different at various vantages within that room, right? Only from outside the room (for example) would the phase relationships remain analogous @ distance.?.

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    Supporting Member Chuck H's Avatar
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    Quote Originally Posted by Mike Sulzer View Post
    Yes, in a room you have many paths between source and listener, and so the phase is very complicated when you add them all up. If the human ear-brain did have high sensitivity to phase, it would be very distracting when you move.
    I think this is what Gnobuddy was getting at then. IMHO it's way too variable to distill into an emulation program (more than I would venture anyway!). But some degree of parametric notch filtering might well be useful for really nailing a room effect and making it tune-able for many circumstances That's as far as I think is practical though. Trying to pinpoint useful, preset EQ parameters for this affect would be a goose chase.
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    Quote Originally Posted by Chuck H View Post
    I think this is what Gnobuddy was getting at then.
    Mainly I was trying to say that I think people often overestimate the importance of phase inside an audio circuit. Back in the days when there was serious study going on of the way we hear sound (a lot of it by Bell Labs), there were many experiments that all came to the conclusion that people really don't hear phase.

    For example, you can take a square wave signal, put it through an appropriate all-pass filter, and it will come out looking nothing like a square wave, because each of the many frequencies in it has been phase-shifted by a different amount. But an all-pass filter has a flat frequency response, so all the frequencies in the original square wave are still present at their original strengths.

    And when listening tests were conducted, guess what? People couldn't hear the difference between the original square wave, and the filtered wave that looked nothing at all like a square (but contained all the same frequencies in the same proportions). They sounded identical.

    This, and so many other experiments, all came to the same conclusion: it's really the frequency response that matters. Get that right, and the accompanying phase seems to be pretty much irrelevant. It might look different on a 'scope or in a Fourier transform, but it will sound the same.

    Which is why I don't think you have any cause for worry about the phase changes that come along for the ride with your speaker-emulation filter. If you get the frequency response to sound the way you want it to sound, that's all that matters!

    As far as I know (and I know very little), simulating room ambience is a very different and far more complex thing. That involves creating dozens or hundreds of fake wall reflections, by time-delaying and adding together dozens of audio signals, usually in the digital domain these days. The amounts of phase involved are huge, thousands of times bigger than the phase shifts that occur inside an amplifier or filter. As I mentioned earlier, 360 degrees worth of phase for every wavelength travelled, so if you take a high frequency and bounce it off a wall many feet away, you get tens of thousands of degrees of phase shift compared to the direct sound. And when you hear both direct and reflected sounds at the same time, then the phase and time differences between them matter, in the sense that your brain hears something going on.

    The people who design studio-grade reverb units know all about this stuff - they've been fooling us into hearing ambient space around recording artists for decades now. I still remember listening to The Eagles for the first time on a Sony Walkman, and loving that huge beautiful airy space that seemed to surround them. That big airy ambience was such a big part of their sound!

    Closer to the original topic of this thread, I have been trying to make a silk purse out of a sow's ear - the short version is that I'm trying to build a guitar amp, on a very tight budget, to give to a friend. The budget dictates that I use a pair of 6.5" woofers that came out of thrift-store boombox speaker cabs, $5 for the pair. They sounded absolutely nasty on my first attempt, but by tinkering with a graphic EQ pedal, I found that putting a notch in the frequency response at 800 Hz took most of the nastiness away. Adding a little gentle bass boost below the notch, and then rolling off the high treble above the guitar range, improved things quite a bit more.

    The story isn't finished yet, and I'm still tweaking things, but essentially, I'm trying to make an emulation filter that makes a contemporary small speaker sound like a guitar speaker. Obviously, only at relatively low volumes, which is how this amp will be used.

    So while Chuck is building a filter to conjure up the sound of a guitar speaker/cab that isn't actually there at all, I'm trying to build a filter that makes a 6.5" boombox woofer sound like a big floppy-coned guitar speaker with a stiff suspension, weeny magnet, limited treble response, and all the other bad design characteristics which happen to be the way we like our guitar speakers.

    -Gnobuddy

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    People don't hear absolute phase, but hit the invert switch on one band of your active crossover and see if you hear it. Invert the connections to the mids on your PA cap or even the horn, and see if you hear it.

    When I ran sound, I used to put up music on the mains and walk across the back of the floor and I could hear the phase cancellations as I walked across the pattern from the two speaker columns. The woofers and tweeters having different patterns.

    Whether those matter to a speaker emulator is another matter.
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    Quote Originally Posted by Enzo View Post
    People don't hear absolute phase, but hit the invert switch on one band of your active crossover and see if you hear it. Invert the connections to the mids on your PA cap or even the horn, and see if you hear it.

    When I ran sound, I used to put up music on the mains and walk across the back of the floor and I could hear the phase cancellations as I walked across the pattern from the two speaker columns. The woofers and tweeters having different patterns.

    Whether those matter to a speaker emulator is another matter.
    When we deviate from a point source, the phase makes cancellations, in the same way standing wave patterns can make a difference between a 'good' sounding room and bad. But from a point source I agree with Gnobuddy that phase is inaudible, having learned of the same studies he mentions. Of course, moving phase (like from a phasor or flanger) is the same as walking through a static sound field and we hear the comb filtering.
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    Quote Originally Posted by Enzo View Post
    People don't hear absolute phase, but hit the invert switch on one band of your active crossover and see if you hear it. Invert the connections to the mids on your PA cap or even the horn, and see if you hear it.

    When I ran sound, I used to put up music on the mains and walk across the back of the floor and I could hear the phase cancellations as I walked across the pattern from the two speaker columns. The woofers and tweeters having different patterns.

    Whether those matter to a speaker emulator is another matter.
    What you are hearing is changes in the amplitude of the frequency response caused by frequency dependent cancellation and reinforcement of multiple sources. You are not directly hearing phase changes.
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    Quote Originally Posted by Mike Sulzer View Post
    What you are hearing is changes in the amplitude of the frequency response caused by frequency dependent cancellation and reinforcement of multiple sources. You are not directly hearing phase changes.
    Exactly. The key words are "multiple sources". I did actualy mention this in an earlier post in this thread (#12), where I said "Phase only really seems to matter when you have two or more drivers simultaneously emitting the same signal, with a phase shift between them. In that particular case, the multiple signals will interfere with each other, and cause peaks and dips in the frequency response."

    Enzo is absolutely right about switching the phase of a tweeter wrt the woofer, of course, and that is a classic example of "two or more drivers simultaneously emitting the same signal, with a phase shift between them"!

    -Gnobuddy

  25. #25
    Supporting Member The Dude's Avatar
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    I've been following without much to add. Interesting discussion. I will say that one of the ways the BBE devices work is to add correction for speaker phase "problems". If a person wanted to actually hear what's being talked about in this thread, a BBE unit would be a good place to start.
    Chuck H likes this.
    “Yeah, well, you know, that’s just, like, your opinion, man.”

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