
Home  Forums  Rules  Articles  The diyAudio Store  Gallery  Blogs  Register  Donations  FAQ  Calendar  Search  Today's Posts  Mark Forums Read  Search 
MultiWay Conventional loudspeakers with crossovers 

Please consider donating to help us continue to serve you.
Ads on/off / Custom Title / More PMs / More album space / Advanced printing & mass image saving 

Thread Tools  Search this Thread 
28th December 2012, 11:00 PM  #111  
diyAudio Member

Quote:
That said, phase response should not be altered by a nongated measurement, as reflexions and other things that occur after the main impulse will not mess up frequency that are already "fully determined" (for lack of a better term) at that point of the impulse. Quote:
It is good though to remove any excess phase to minimize the work that has to be done to obtain this perfect impulse, as it will also make the best use of the available taps in the convolution engine. The other reason to aim for the minimal phase reponse in your measurement is because it will be easier to correct to boot, and should mimic the textbook theoretical behavior of the driver and filters you are using. Quote:
I would stay away from the "inverse impulse response" type of correction, as it requires a lot of care to be done properly: a lot of measurements and averaging to avoid correcting things that should not be corrected, which tends to make the whole "automatic correction" thing kind of a moot point... (or a least a lot less appealing than it seems). You can try to generate this "reverse impulse" with DRCFIR, as it will already take care of a lot of potential pitfalls for you (frequency dependent windows, "intelligent" corrections, target curves, etc.), but you will need a good set of measurements, and as DRCFIR only takes one measurement as input, you will probably have to do an averaging of several good measurements (spatial averaging). And of course to be able to do this averaging all the impulses will need to have the same offset, so... you better stick with minimum phase for all your measurements Last edited by pos; 28th December 2012 at 11:03 PM. 

28th December 2012, 11:10 PM  #112 
diyAudio Member
Join Date: Nov 2011
Location: Cooktown, Oz


29th December 2012, 12:58 AM  #113 
diyAudio Member

This is probably the correct phase response in a theoretical point of view, but how to determine the frequency and Q of that LP filter?
Looking for a phase ~0° at the nyquist frequency seems to be the easiest way to look at it for corrections. 
29th December 2012, 01:23 AM  #114  
diyAudio Member
Join Date: Aug 2004
Location: US

Quote:
The next issue is whether, in a multiway system, you linearize each driver+ filter individually or first get the response of the entire system and linearize the system phase. There are advantages and disadvantages to each approach. If you do each driver + filter individually then in each case the correction is the inverse of the minimum phase plus a delay where the delay can be positive or negative or zero. If you linearize the system the system phase is just some nonlinear, and generally non minimum phase which may or may not have a linear phase component in it. In simple terms the correction whether for the system of individual band passes is just an all pass filter with phase that is the inverse of that to be linearized. Once you know what you want to linearize it's pretty trivial to construct the impulse needed.
__________________
John k.... Music and Design NaO Dipole Loudspeakers. 

29th December 2012, 01:52 AM  #115  
diyAudio Member
Join Date: Nov 2011
Location: Cooktown, Oz

Quote:
Phase at Nyquist is BY DEFINITION = 0°. So for an "accurate" Fourier block to represent your system, it should have a phase which reaches some multiple of 180° before it reaches Nyquist. What algorithm are you using for your FIR optimization? I use a very crude version of Remes exchange. 

29th December 2012, 09:19 AM  #116 
diyAudio Member
Join Date: Jan 2006
Location: grenoble

@ coppertop
a capture of minimal phase for a tweeter. simulated with rephase> export to REW>generate minimal phase. Last edited by thierry38efd; 29th December 2012 at 09:25 AM. 
29th December 2012, 11:28 AM  #117  
diyAudio Member
Join Date: Aug 2004
Location: US

Quote:
The correct minimum phase for that response should look like this assuming the response above 20k Hz continued to roll off at about 2nd order. If I assume the high frequency roll off is 6db the MP looks like this: Lastly, to get a phase response than looks something like yours I need to assume that the response goes flat at DC and above 20k Hz:
__________________
John k.... Music and Design NaO Dipole Loudspeakers. Last edited by john k...; 29th December 2012 at 11:45 AM. 

29th December 2012, 11:57 AM  #118 
diyAudio Member
Join Date: Jan 2006
Location: grenoble

ok.curve in low frequency isn't correct,because of the FIR impulse resolution/ripple in this case.(it's a fast way to create a curve with linear phase file and use it in another software ).
purpose was to show what a minimal phase looks like in the concerned band.(ie 500 Hz>20000) i was thinking minimal phase of a function was (delta spl/delta frequency). i believe this is how REW extract minimal phase of a frequency curve.(Hillbert transform) my mic calibration and some example found around here. Last edited by thierry38efd; 29th December 2012 at 12:10 PM. 
29th December 2012, 12:15 PM  #119  
diyAudio Member

Quote:
Quote:
It is similar to Phase Arbitrator in this regard. So for example you can take the specsheet of a JBL LSR25p and use the acoustical slopes specified (acoustical LR24 at 2.3khz) as well as the BR tuning (they even specify the whole HP function of the system). This is a special example because not that many brand specify that much, but you can almost always at least guess the acoustical crossover points and slopes, as well as the BR tuning. After you dial these figures into rephase (filter linearization tab) you should already get something good as far as phase is concerned. Then the user can use measurement to fine tune the correction (either by adjusting the filter linearization configuration, or by using the phase paragraphic EQ). It has been my experience that when the speaker is well behaved only minimal adjustments have to be done. So when doing the measurement the easiest way of doing the correction with rephase is to try to "see" the theoretical crossovers you were expecting (from the spec). So having a 0° (or 180° multiple) target at nyquist is the way to go (unless you know the exact LP function of the system up high, and also the one of your measurement system...). Another reason to aim for a given "target" (impulse offset) instead of just linearize an arbitrary measurement is to be able to use multiple measurements (or averages) to do the linearization: they need to have the same offset (with regard to the impulse peak) to be comparable (ie to be able to average them, or use the same correction on any of them). Quote:
If you set the centering to "int" (it was the only option in rephase prior to version 0.9.0) you cannot always get a perfect 180° multiple up high and can end up with ripples (at the nyquist freq), especially if you do not use a smooth window function (that is why the "complex" window function was implemented: it is a rectangular window, mixed with a hann window for frequencies above nyquists/2...) Quote:
Of course when only doing linear phase operations (filtering or eq) the impulse is always symmetrical so this optimization is useless, but when doing phase corrections it starts to be quite useful. For example if you only do minimum phase corrections you will end up with the peak at the left of the final impulse (window on the right of the double impulse), whereas if you correct an existing minimum phase system to linear phase you will end up with the peak at the right of the impulse (window of the left of the double impulse). And if you do both phase linearization (to correct the natural behavior of a driver) and linear phase filtering (to do the actual crossover) you end up with the peak at the "optimal" place... The chosen window function is applied asymmetrically around the peak on the final impulse. The second optimization is iterative and takes place afterward: it is just trying to modify the target amplitude curve so that the result gets closer to the initial target curve. So at each iteration step the ratio between the result and the target is calculated at each point of the target curve, and the result is multiplied by 1.1 (to speed things up) and the amplitude value of the point is multiplied by this value. So the result curve should get closer and closer to the initial target... When the "minimal" optimization is chosen there is only one optimization step, and the ratio is of course not multiplied by 1.1 I found afterward that this iterative optimization was already used by Rainer Thaden in the Four Audio HD2 processor: http://www.studitech.ru/resque/manua...ES32_rev5.pdf How do you apply that Remez exchange algorithm? 

29th December 2012, 01:02 PM  #120 
diyAudio Member
Join Date: Aug 2004
Location: US

Minimum phase extraction is a complicated thing. Most measurement codes use an approach based on the Real Cepstrum and discrete FFT/IFFT. The problem with those approaches is that they do not typically give accurate results at the frequency extremes. That is because of the periodic nature of the FFt. If you look at Bode's book, Network Analysis and Feedback Amplifier Design, you will see that the relationship between amplitude and phase for minimum phase systems involved an integral that extends fro DC to infinite frequency. With the discrete approach, obviously we don't have any information above the Nyquist frequency. But, if the high frequency roll off of the device is well established a couple of octaves below the Nyquist frequency good result can be obtained. The problem comes in when the response of the device (like a tweeter) extends past the Nyquist frequency.
The thing about the integral approach is that it tells you something very important. It tells you that while the phase at any frequency depends on the amplitude from DC to infinity, it also tells you that the contribution to the phase at a given frequency weights the amplitudes nearest that frequency most heavily. And it also tells you what the phase is once the slope is established in the roll off region. For example, if the response has a constant slope of Ndb/octave the slope well be 15 x N degrees. So you know just by looking at the response of a band pass filter, for example, what the phase asymptotes will be. For example, a band pass with a 4th order high pass and a 3rd order low pass characteristic will have 360 degrees phase at DC (24 x 15) and 270 degrees at infinity (18 x 15). (These are unwrapped phase).
__________________
John k.... Music and Design NaO Dipole Loudspeakers. 
Thread Tools  Search this Thread 


Similar Threads  
Thread  Thread Starter  Forum  Replies  Last Post 
FIR linear phase plugin for MiniDSP?  diyjb01  miniDSP  13  7th February 2014 01:24 AM 
FIR filter design tool for Loudspeaker magnitude equalization  ttmusic  Software Tools  3  24th May 2013 08:30 PM 
FIR Filtering experiences  Olombo  PC Based  8  10th February 2013 03:45 PM 
AVX based FIR VST, crossover / EQ / DRC and delay  KOON3876  PC Based  97  26th November 2012 07:18 AM 
Phase EQ using FIR filters  Grasso  MultiWay  2  2nd July 2003 10:37 PM 
New To Site?  Need Help? 