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Old 29th December 2012, 01:11 PM   #121
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I though I would throw up this old web page of mine from when I was involved with the Bodzio UE. I'm nor longer involved. This was an early version. The newer versions use a CAD like circuit layout to define the target functions.

UE Open baffle
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Old 29th December 2012, 02:26 PM   #122
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Quote:
Originally Posted by CopperTop View Post
Too late to edit my previous post. I was going to add:

I still haven't got my head around phase vs. delay when it comes to 'correction'. If there's a phase deviation of >360 degrees (which constitutes a delay..?) I'm assuming that it's not possible to correct it by phase only, and maintain time domain coherency of a transient - which I had thought was the main aim of the exercise, apart from matching drivers' phase in the crossover region.

Do real world drivers exhibit a phase characteristic and also a delay characteristic i.e. a pulse can be set in motion that doesn't emerge at the diaphragm until a fixed propagation delay has passed?

Is there an upper frequency limit to phase correction?
Think of unwrapped phase delay. Smooth slope without excessive number of inflection points and lumpiness in between is first objective. Removing main slope is second.

Phase becomes less and less important with increasing frequency. Sure, flattening it out to Nyquist frequency is neat, but just considering content in natural sources; How much evolutionary survival value is there in phase information above 2kHz? Localizing is intensity driven up high. Getting tweeter to integrate with range below as single source is primary effort.

Measurement to diaphragm surface doesn't help. JohnK approach of referencing to baffle is direct measurement approach to excess phase. Time of flight differences is real target.

If measurement system is stable, back to back measurements yield identical timing of IR location in recovered waveform. This is easy to test. When this is case, switching from tweeter to woofer measurement preserves exact time of flight difference between the two drivers.

Some measurement systems/software behave well, some don't.

Use of multiple sweeps with a set number of sample in between allows tweeter to be measured, muted, then woofer unmuted and measured in single measurement. Raw sweeps are split based on sweep length and inserted sample length. Results show perfect relative orientation in time.

Waveforms are adjusted to same amplitude at crossover frequency. Timing is tweaked so phases match at crossover frequency. If phase slopes match extended overlap occurs, otherwise phase slopes intersect. Differing slopes may be differences in drivers, and sometimes indicates a driver needs leads reversed (and more tweaking).

Measurements need to be done at distance where baffle effects become inconsequential.

Broadband minimum phase/causal impulse systems must have group delay, and thus phase response with slope. All frequencies start at zero amplitude at t=0.

Although virtually all natural sound sources have causal behavior, and speakers have response, information transmission is continuous system, and is best treated as such. Bandwidth is bandwidth, not IIR bandwidth or FIR bandwidth.

Filters viewed as waveforms may lead many to hear what they are looking at. Much as many great speakers sound great until room measurement is taken, and suddenly that floor/ceiling mode low frequency behavior at listening position sounds just terrible.

High order crossover filters sound bad regardless of IIR/FIR nature due to underlying design choices. Causal filter's ringing is better masked to human perception by precedence effect, but high order filters have loads of extended ringing, that when misaligned with conjugate leads to descriptions such as "sounds phasey".

Addition of 100u capacitor for tweeter measurement is adding several degrees of phase change. Limiting power, and or sweep range is best.

Regards,

Andrew
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Old 29th December 2012, 02:28 PM   #123
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Quote:
Originally Posted by john k... View Post

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).
thanks for the explanations.
it's useful to understand limits of discrete filtering/EQ.

may we have not to considering the high frequency phase resulting.
it's hard to hear a difference between linear phase and minimum phase in the lows,so in the highs...
otherwise,this is a good exercise,it 's free to linearize,using FIR EQ/Xover.
even if there's no audible improvement with low group delay.
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Old 29th December 2012, 02:41 PM   #124
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Here is a section form the manual I wrote for the UE describing the measurement procedure and why a single position with fix amount of excess delay removed is preferred.

Click the image to open in full size.
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Old 29th December 2012, 02:42 PM   #125
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Quote:
Originally Posted by Barleywater View Post
Think of unwrapped phase delay. Smooth slope without excessive number of inflection points and lumpiness in between is first objective. Removing main slope is second.

Phase becomes less and less important with increasing frequency. Sure, flattening it out to Nyquist frequency is neat, but just considering content in natural sources; How much evolutionary survival value is there in phase information above 2kHz? Localizing is intensity driven up high. Getting tweeter to integrate with range below as single source is primary effort.

Measurement to diaphragm surface doesn't help. JohnK approach of referencing to baffle is direct measurement approach to excess phase. Time of flight differences is real target.

If measurement system is stable, back to back measurements yield identical timing of IR location in recovered waveform. This is easy to test. When this is case, switching from tweeter to woofer measurement preserves exact time of flight difference between the two drivers.

Some measurement systems/software behave well, some don't.

Use of multiple sweeps with a set number of sample in between allows tweeter to be measured, muted, then woofer unmuted and measured in single measurement. Raw sweeps are split based on sweep length and inserted sample length. Results show perfect relative orientation in time.

Waveforms are adjusted to same amplitude at crossover frequency. Timing is tweaked so phases match at crossover frequency. If phase slopes match extended overlap occurs, otherwise phase slopes intersect. Differing slopes may be differences in drivers, and sometimes indicates a driver needs leads reversed (and more tweaking).

Measurements need to be done at distance where baffle effects become inconsequential.

Broadband minimum phase/causal impulse systems must have group delay, and thus phase response with slope. All frequencies start at zero amplitude at t=0.

Although virtually all natural sound sources have causal behavior, and speakers have response, information transmission is continuous system, and is best treated as such. Bandwidth is bandwidth, not IIR bandwidth or FIR bandwidth.

Filters viewed as waveforms may lead many to hear what they are looking at. Much as many great speakers sound great until room measurement is taken, and suddenly that floor/ceiling mode low frequency behavior at listening position sounds just terrible.

High order crossover filters sound bad regardless of IIR/FIR nature due to underlying design choices. Causal filter's ringing is better masked to human perception by precedence effect, but high order filters have loads of extended ringing, that when misaligned with conjugate leads to descriptions such as "sounds phasey".

Addition of 100u capacitor for tweeter measurement is adding several degrees of phase change. Limiting power, and or sweep range is best.
All great stuff, thanks. I think I understand the unwrapped phase thing, now. I was worried that each FFT coefficient can only be phase-shifted by +/- 180 degrees, so how could I correct for >360 degrees of shift, but of course it's the cumulative effect of the multiple 'wraps' that produces the delay i.e. the FFT of a delayed impulse with its multiple phase rotations can simply be reversed to give the original delayed impulse again. It's not ambiguous at all.

When you say

Quote:
Measurements need to be done at distance where baffle effects become inconsequential.
does that mean very close, or far?

If using REW to make impulse response measurements, should I be using its minimum phase export routinely, rather than attempting to do anything clever myself?
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Old 29th December 2012, 03:12 PM   #126
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Originally Posted by pos View Post
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.
Could the phase error also be taken into account? What currently happens when trying to synthesize constrained length filters with parametric or graphic phase EQ is the passband phase response tends to gets stuck around 45 degrees from the target curve. My experience is the threshold of phase audibility is around 10 degrees so, to be usuable, the filters should ideally within 5 degrees of target. Since the length is constrained by hardware adding more taps to improve the phase accuracy of the initial solution isn't an option---performance is governed by how much phase accuracy can the optimizer can wring out of the available number of taps.

Also, it'd be handy if the range of the phase axis in the graph could be changed like that of the magnitude axis. If you're trying to see if a phase error is less than five degrees having the axis cover 360 degrees is too much.

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Originally Posted by pos View Post
How do you apply that Remez exchange algorithm?
Not quite sure what you're asking here; Remez exchange is well documented on the web and in pretty much any intro DSP text so I assume you're familiar with its equiripple innards and lack of phase awareness as usually defined---it's usually used with linear phase synthesis, after all. I'd also argue the core operations of Remez, conjugate gradient, and other iterative solver methods are not fundamentally different that what's already implemented in rePhase, though different flavours of the math offer different convergence behaviours. My experience leads me to suspect a change of algorithm would improve rePhase's current magnitude performance.

A while back I looked into phase aware optimization a bit. As so much of FIR synthesis is linear phase there's not much out there. I'd coded conjugate gradient in complex math around a decade ago and had pretty good results with both magnitude and phase convergence for the problems I was solving with it. So I was thinking of starting by giving it a try for filter synthesis.
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Old 29th December 2012, 03:30 PM   #127
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Quote:
Originally Posted by pos View Post
Any measurement can be corrected (either manualy using rephase, or automatically using DRC-FIR or UE for example), and after correction you should get a "perfect" impulse response
I've done this by hand in HOLMImpulse using smoothing and a target and also had it done in PORC. The results were similar, tho I can't say that either sounded any better than the non-corrected response, just different. (for my system)

Quote:
Originally Posted by Barleywater View Post
Some measurement systems/software behave well, some don't.
I've had very good results with HOLMIpulse. Time can be locked and seems to stay that way with few problems. I've done a number of tests and found it to be stable as long as the computer is. I have one laptop that is not because of interrupts.
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Old 29th December 2012, 04:22 PM   #128
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1" dome tweeter has potential for omnidirectional behavior to several kHz, so faceplate edges and edges of flat baffle "see" tweeter in crossover region, and baffle edge becomes delayed source. Measurements should be taken at a distance on the order of 6x the effective radiation aperture to get baffle step, or measure real close to swamp out reflections, and add totally synthetic baffle step correction.

This is tempered against ratio of direct to reflected energy, and ability to gate out, or potentially use. Correcting subs below 2nd mode frequencies of room allows great latitude in microphone placement, but reflections much above 200Hz (In my experience) introduce artifacts that range from hard to pick out, up to sounds not unlike whistling into a fan blade when using direct inversion of IR as correction.

Many using DRC find this, and by beating out artifacts are also left with little effective correction.

For Pluto type speaker good baseline measurement is possible at 9" for direct inversion to create correction filters for both drivers. These may be driven with crossover and single correction applied; or may be individually measured from same point, responses inverted, and desired crossovers applied to correction filters for result.


Synthesis of correction is only as good as synthesis method is in recreating real response.
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Old 29th December 2012, 08:25 PM   #129
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Originally Posted by pos View Post
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_rev-5.pdf

How do you apply that Remez exchange algorithm?
I haven't found strict Remez exchange good for anything but very high slope 'classical' functions. It's no good for the type of arbitrary response corrections required in speakers & mikes.

My bastardised method has much in common with Thaden & your method except I use the 'phase' info. I first got the idea from either Numerical Recipes or Oppenheim & Schafer.

Don't have access to books. I've been a beach bum for well over a decade and only emerged from the bush in 2005 when the magic internet reached Cooktown, Oz.
_____________

Gotta declare my prejudices if they are not already clear from the 2 AES papers.

IMHO, the correct way to do digital EQ is to equalise the Minimum Phase part using IIRs as in my 2008 paper. Then if you want the phase correcting (I don't in most cases), pick and choose the exact parts to EQ and use a FIR.

This is the approach used by Greenfield - Efficient Filter Design for Loudspeaker Equalization He's also got some papers on the evils of using Linear Phase xovers.

What's exciting for me is this millenium, we now have the computing power to do speaker+room. But the optimum strategy for this is yet to be dreamt up. This mythical beast would take my "surround sound system is a speaker" to the next level.

As in ALL EQ, what you DON'T EQ is as .. if not more important as what you do EQ.

Last edited by kgrlee; 29th December 2012 at 08:41 PM.
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Old 29th December 2012, 11:05 PM   #130
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Quote:
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As it is now, to see the effect of a correction you can do the convolution inside HOLMImpulse, using the "C=A*B" manipulation : you put your measurement in one slot, the correction impulse in another, and you generate the result in the C slot.
When I do this, I get a perfectly flat phase plot, but no amplitude! Never seen a HOLM plot with no amplitude trace. What am I doing wrong?
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