• WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

Minimum Phase

UliBru

Active Member
Technical Expert
Joined
Jul 10, 2019
Messages
124
Likes
338
A linear time-invariant system reacts on a Dirac pulse with its own pulse response. This pulse response fully describes the characteristics of the system under test.
Now the pulse response can be investigated. So e.g. a Fourier transform will give us information about the frequency and phase response. We can consider the pulse response as a filter and feed it with proper signals. A step input signal thus shows us the step response. There are many ways of investigation. Each one shows some detail about the characteristics.

Now we can e.g. derive the frequency response by the Fourier transform and think about a pulse which creates this response in the shortest time and with minimal phase changes. The answer is - not surprising - the minimumphase pulse response. It has the important feature that it is causal. It describes the reaction of the system to an input signal. The system will not react before the input event happens. A loudspeaker will not play music before it gets fed with it. So physical systems usually react causal and minimumphase.

But there is another feature in the game: delay. A sound wave e.g. takes its time to arrive. And thus many things can happen in between and add or deform the pulse response. That's why e.g. the pulse response of a multiway speaker is no longer minimumphase. The drivers have their individual timings and thus the added characteristics does not show the shortest time to create the overall frequency response.

Now we can describe any given NMP pulse response IR as a combination of two underlying pulse responses MP and EP. These sub-pulses MP and EP are connected by convolution and so we can write
NMP = MP * EP (* = convolution)
MP is now the minimumphase pulse response and EP is the so-called excessphase pulse response. We have already talked about minimumphase. The excessphase EP is nothing else than an allpass. It's frequency response is perfectly flat, so every sinewave will pass the allpass filter without change of amplitude. But the allpass allows to change the timing or phase. It simply delays frequencies.
The convolution of MP and EP leads to an indefinite number of resulting impulse responses. We find usually three terms in discussion: minimumphase, linearphase, maximumphase.
All of them are still a result of MP*EP. In case of minimumphase both MP and EP are minimumphase. In case of linearphase EP is cancelling the phase changes of MP. There is no phase change to achieve a desired frequency response. In case of maximumphase EP creates the maximum phase delay. In general MP*EP can be described as mixedphase.

There are some important points regarding MP and EP:
- the inverse of MP is again minimumphase. The frequency response can be seen as mirrored along the frequency axis. It is possible to derive MP from its frequency response.
- the "inverse" of EP is the time-reversal of EP
We can apply these actions when we manage to split IR into MP and EP properly.

And if we like to correct MP and EP we "just" have to apply these two inversion operations. It sounds simple but of course the real world will limit this. Think about the frequency response of a tweeter and invert it. You can see you need to boost a lot in the bass area to get a flat tweeter response. But the tweeter will certainly not survive it.
Think about a comb filter where two sinewaves of same frequencies but 180° phase shift will perfectly cancel to zero. There is no way to boost zero.
These examples are described by the term ill-conditioned. And we can and do find many ill conditions when we try to measure and correct something. The only way is to make cutbacks and reduce the requirements for an ideal correction. The art of correction lies in the treatment of ill conditions.

Just my humble opinion.
 
Last edited:

René - Acculution.com

Senior Member
Technical Expert
Joined
May 1, 2021
Messages
427
Likes
1,309
It should be added that a physical system is not inherently MP or NMP, as several transfer functions can be associated with a single system, and some of these may be MP while others are NMP, depending on the source(s) and where the output is taken. So when for example stating that a particular driver is MP (or not), there is an assumption about a transfer function from the output pressure at some distance, taking out the phase shfit from the travel time, back to the input voltage, whereas one could create other transfer functions regarding the displacement at specific points on the cone, which could be NMP, as is the case for some flexible systems.
 

dasdoing

Major Contributor
Joined
May 20, 2020
Messages
4,301
Likes
2,774
Location
Salvador-Bahia-Brasil
We can apply these actions when we manage to split IR into MP and EP properly.

this implies that the spilt is not trivial?
one would think that you can create the expected phase response from the measured FR and define the excess as EP....and afaik REW does it this way.
is it more complicated than this?
 

UliBru

Active Member
Technical Expert
Joined
Jul 10, 2019
Messages
124
Likes
338
this implies that the spilt is not trivial?
one would think that you can create the expected phase response from the measured FR and define the excess as EP....and afaik REW does it this way.
is it more complicated than this?
If
NMP = MP * EP
then
EP = NMP / MP (with / = deconvolution)

It is this simple without ill-conditioning
 

gnarly

Major Contributor
Joined
Jun 15, 2021
Messages
1,037
Likes
1,474
And if we like to correct MP and EP we "just" have to apply these two inversion operations. It sounds simple but of course the real world will limit this. Think about the frequency response of a tweeter and invert it. You can see you need to boost a lot in the bass area to get a flat tweeter response. But the tweeter will certainly not survive it.
Think about a comb filter where two sinewaves of same frequencies but 180° phase shift will perfectly cancel to zero. There is no way to boost zero.
These examples are described by the term ill-conditioned. And we can and do find many ill conditions when we try to measure and correct something. The only way is to make cutbacks and reduce the requirements for an ideal correction. The art of correction lies in the treatment of ill conditions.
Hi, I've been using impulse inversion for multi-way speakers, but so far only at the driver level with each driver section getting its own individual FIR file.

Using your example of a tweeter, my approach has been to flatten its magnitude with minimum phase EQs way below its intended passband, and nullify the driver destroying bass gain with a steep complementary linear phase crossover.
It's been pretty easy to get drivers' acoustic responses to match textbook Linkwitz-Riley xovers magnitude for whatever safe order I choose, and with flat phase.
(at least on axis, quasi-anechoic. I then measure the FIR corrected drivers polars to assess the degree of valid correctability.)
Do you see any caveats with this technique?

With regards to your comb filter example....yeppa, I've found impulse inversion can't fix acoustic issues from multiple sources, even just at the simple level on speaker only.
Heck, even from a single driver, and its cabinet interactions.

I've yet to venture into room correction, and discussions like these are of high interest, because I want to be able to dissect at the room response level, the correctable from the uncorrectable, as well their extents of correctability.
 

UliBru

Active Member
Technical Expert
Joined
Jul 10, 2019
Messages
124
Likes
338
Do you see any caveats with this technique?
No, as you are thinking about the "conditions" and what you can do to avoid problems :) And there are many ways to skin a cat.
 

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
679
Likes
499
Location
Germany
It is this simple without ill-conditioning
And the perfect EP inversion filter with "infinite" taps would violate causality. Another ill condition of the nature.
 

UliBru

Active Member
Technical Expert
Joined
Jul 10, 2019
Messages
124
Likes
338
And the perfect EP inversion filter with "infinite" taps would violate causality. Another ill condition of the nature.
Inverting the EP means to reverse the recorded or calculated EP pulse. As the pulse usually is not of infinite length it is no problem to process it. You can easily reverse a pulse of 10 million taps. But of course a long pulse means a long time. With this 'silly' example at 96 kHz the pulse has a 104 s duration time. And of course it does not make sense to start a speaker playback pre-correcting directly the room behaviour which happens 104 s later with a playback starting without correction. Welcome to reality.
 

dasdoing

Major Contributor
Joined
May 20, 2020
Messages
4,301
Likes
2,774
Location
Salvador-Bahia-Brasil
Is the perfect (room response) inversion even possible?
sure, we could theoraticly phase invert all the reflections, but this inverse filter will go through the system and recieve those reflections also.
 

UliBru

Active Member
Technical Expert
Joined
Jul 10, 2019
Messages
124
Likes
338
Imagine you have a perfect room with perfect speakers. The recorded pulse response is a Dirac pulse

t1.png


Now the left ear receives the right speaker signal with some delay (about 7 cm more distance) and a bit attenuated (e.g. -1 dB). With right ear it is similar, left left speaker is delayed.

t2.png


So the ear receives the direct signal and the reflection as sum

t3.png


The frequency response looks like this, it shows a typical comb filter

m1.png


As the summed pulse response is minimumphase we can invert the magnitude response and derive a correction filter

m2.png


t4.png


The correction shows the required pulses as a descending sequence of anti-pulses. The convolution of the sum pulse with the correction results in a perfect Dirac pulse and flat magnitude response

t5.png
 

UliBru

Active Member
Technical Expert
Joined
Jul 10, 2019
Messages
124
Likes
338
m3.png


It is perfect, right?

But now let's assume the head is moved 1 cm. So the left ear receives the right speaker just a bit later.
So we get a slight different sum and comb filter.
Unfortunately we apply the previous correction filter on the new situation.
The resulting corrected pulse response (convolution of actual room/speaker pulse response with previous correction is now

t6.png


and the frequency response is

m4.png


So obviously now the perfect correction looks a little non-perfect.

I hope this little example demonstrates that all small-talk about a PERFECT correction is nonsense. A room correction should not try to correct reflections.
 
Last edited:

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
679
Likes
499
Location
Germany
A room correction should not try to correct reflections.
I will need to add to that "for wavelengths above the room transient and definetely below the distance between ears!"
 

dasdoing

Major Contributor
Joined
May 20, 2020
Messages
4,301
Likes
2,774
Location
Salvador-Bahia-Brasil
I will need to add to that "for wavelengths above the room transient and definetely below the distance between ears!"

man, I tried so many things in the last months.
the best you can do is drc the first wave and aprouch linear phase response, or minimum phase response (for the old-school sound).
otherwise you will always end up with artifacts
 

ernestcarl

Major Contributor
Joined
Sep 4, 2019
Messages
3,113
Likes
2,330
Location
Canada
A room correction should not try to correct reflections.

What are your thoughts on recently popular "active room correction" using multiple speakers distributed around the room? Does it not successfully negate/correct/reduce the effect of some of these (generally bass only) reflections? The bulk of the work seems to be via "active absorption".

Michael Gerzon wrote as early as 1991:

"... once one has multiple loudspeakers dotted around the room, as in home cinema or HDTV applications, the additional loudspeakers provide the possibility of controlling low frequency room colouration by using the principles of “active absorption”, where the room response is used to provide electronically assisted absorption by the speakers of room resonances. However, active absorption techniques are not used in the first generation of room/speaker equalisation products, and in practice are unlikely to be applicable to use with just two loudspeakers, since these do not give adequate absorption of resonances."


Some pro audio companies may be at the beginning stages of implementing this idea more or less to a certain degree:

Fulcrum Acoustic Immersive
1692383669716.png


The Venueflex™ processor continually samples the subwoofer drive signals from an installed audio system, and produces complementary and adaptive signals to drive numerous low-frequency control loudspeakers distributed around the room. This creates tighter, more controlled sub-bass response for an improved audio experience.

*Well, it's likely more than simple "active absorption" in Fulcrum's case with their large venue applications.


Though I have yet to see/hear it in action.
 
Last edited:
  • Like
Reactions: OCA

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
679
Likes
499
Location
Germany
man, I tried so many things in the last months.
the best you can do is drc the first wave and aprouch linear phase response, or minimum phase response (for the old-school sound).
otherwise you will always end up with artifacts
Dive into the cruel world of 2nd order allpass filters and you can lose a lot more time ;)
 

dasdoing

Major Contributor
Joined
May 20, 2020
Messages
4,301
Likes
2,774
Location
Salvador-Bahia-Brasil
Dive into the cruel world of 2nd order allpass filters and you can lose a lot more time ;)

I did man. And it's a very intresting subject. But I can't forward these delays here with them without creating pre-delay:

1692404387629.png
 

Attachments

  • 1692404078405.png
    1692404078405.png
    606 KB · Views: 41

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
679
Likes
499
Location
Germany
I did man. And it's a very intresting subject. But I can't forward these delays here with them without creating pre-delay:

View attachment 306611

the left part is a FIR highh pass btw
Did you try simultanoeus forward and backward in time 2nd order allpass with different bandwiths on the same frequency? (or removing your curtains or jalousies from the sides for that matter)
 

dasdoing

Major Contributor
Joined
May 20, 2020
Messages
4,301
Likes
2,774
Location
Salvador-Bahia-Brasil
Did you try simultanoeus forward and backward in time 2nd order allpass with different bandwiths on the same frequency? (or removing your curtains or jalousies from the sides for that matter)

the thing is, in order to move these delays between 200Hz and 500Hz forward the correction filter will have necessarily a "spiky" front running shape. And in my experience these will always be audible. If these delays would be more broadband, they would be correctible.
btw, my first reflection points are treated, BUT because my room is small the mains are close to the wall. So these problems are caused by the close to wall position.
I don't understand what you mean by "forward and backward".
 

dasdoing

Major Contributor
Joined
May 20, 2020
Messages
4,301
Likes
2,774
Location
Salvador-Bahia-Brasil
btw, I skipped your last youtube video a lot....and I will still study it.
it's just that I wanted to finalize my own study which I have been working on for literally dozens of hours....if not more.
I have adopted your minimum phase mic correction correction btw. I willl reply to the other 2 topics we are discussing.
 
  • Like
Reactions: OCA

OCA

Addicted to Fun and Learning
Forum Donor
Joined
Feb 2, 2020
Messages
679
Likes
499
Location
Germany
the thing is, in order to move these delays between 200Hz and 500Hz forward the correction filter will have necessarily a "spiky" front running shape. And in my experience these will always be audible. If these delays would be more broadband, they would be correctible.
btw, my first reflection points are treated, BUT because my room is small the mains are close to the wall. So these problems are caused by the close to wall position.
I don't understand what you mean by "forward and backward".
"forward and backward"
"in time"!

These babies:

1692432288348.png


But "use with caution" as the author reminds :)
 
Top Bottom