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Multi-Way Conventional loudspeakers with crossovers

rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
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Old 19th February 2015, 03:18 PM   #621
pos is offline pos  Europe
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Quote:
Originally Posted by jtalden View Post
I have used the Complex window for several of my setups, but haven't investigated the choices deeply. With your intension to drop Complex, I finally did a more extensive trial of the other options. As it turns out the best fit for my situation is the Bartlett-Hann window. I would like to see that one stay, but I can probably get acceptable results with one of the remaining windows if you do remove it.
I have other windowing algorithms implementations that I would like to add.
Can you share one of your rephase files so I can check if one of those would fare better than the bartlett-hann one?
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2019-01-16: rePhase 1.4.3
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Old 19th February 2015, 03:23 PM   #622
pos is offline pos  Europe
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Originally Posted by bambadoo View Post
Possible to implement:
kaiser-bessel ?
Parks–McClellan ?
I plan to add Chebychev window with variable ripple floor, but I have to find a way to make it faster than it is now...
This is similar to the Kaiser window, and has some nice properties.

Parks–McClellan is not a window but a method to build filters, and rephase has its own
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Old 19th February 2015, 07:29 PM   #623
jtalden is offline jtalden  United States
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Quote:
Originally Posted by pos View Post
I have other windowing algorithms implementations that I would like to add.
Can you share one of your rephase files so I can check if one of those would fare better than the bartlett-hann one?
Here is the file. I was using "Bypass" mode to assure the SPL is smooth and flat at Zero. This was my main focus. Some minor deviation of phase at very LF is not as big a concern for me, but I looked to make that as good a possible also.

2015-02-17rePhase44kBartlett-Hann.zip
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Old 19th February 2015, 08:13 PM   #624
pos is offline pos  Europe
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Thanks

It looks like hann window also gives good results with your correction.
Also try to use a higher FFT size, like 65536, so you (and the iterative optimization process) get a more accurate view of what is going on down low.
Also try energy centering is this case: this will give you a better phase matching.

Anyway, I will keep your correction to check the new algorithms. Thank you.

As a side note, are you sure you need that much phase correction down low?
Measurements are often doubtful and almost meaningless at these frequencies, especially in domestic environment, so it is often better to stick with theoretical corrections at these frequencies (BR, etc.), or live them alone altogether.
Also, regarding in band phase correction, have you tried using some minimum-phase EQ to linearize the amplitude, and see the effect on phase?
I plan on adding an "excess phase" visualization in a future version, and use that as the preferred phase EQ target...
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Old 19th February 2015, 11:12 PM   #625
jtalden is offline jtalden  United States
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pos, thanks for the comments.

Quote:
Originally Posted by pos View Post
It looks like hann window also gives good results with your correction.
Also try to use a higher FFT size, like 65536, so you (and the iterative optimization process) get a more accurate view of what is going on down low.
Also try energy centering is this case: this will give you a better phase matching.
Yes, I played with all those options. I also found that Hann would be a good option. Go ahead and drop Bartlett-Hann, I am not really opposed to that.

Quote:
As a side note, are you sure you need that much phase correction down low?
Measurements are often doubtful and almost meaningless at these frequencies, especially in domestic environment, so it is often better to stick with theoretical corrections at these frequencies (BR, etc.), or live them alone altogether.
I am reasonably confident my LF phase rise is accurate for my setup. It includes the phase correction for all my measurement chain devices; Audio Interface, AV Pre/Pro, DCX2496, P-Amp, and mic preamp. So the measurements should be representative of the true speaker response. I have a B-24 HP filter at 20Hz set in the DCX so that adds to the ported SWs. My measurements are set to start at 12Hz so I don’t pay much attention below that.

In my case the It would seem more convenient if RePhase just allowed stacking random filter orders at various freq to best match the phase rollup. The choices offered are not very easy to fit to my real world situation as you saw. I defer to you and others needs however as you are probably the more representative of common usage.

Quote:
Also, regarding in band phase correction, have you tried using some minimum-phase EQ to linearize the amplitude, and see the effect on phase?
I plan on adding an "excess phase" visualization in a future version, and use that as the preferred phase EQ target...
Regardless of my choice of house curve would I not want the direct sound phase to be linear at my LP? My logic has been that Irrespective of my choice of SPL it is still correct to have all freqs arrive at the same time. I suppose if my tapered house curve causes REW/Holm to report phase rotation that is in fact not there then this may be the wrong approach? I am not really sure of my logic on this point.

The SPL of my 5 identical speakers are EQed in the DCX with mic at the ~4m LP. EQ is done as well as I believe is appropriate given the DCX capabilities. The phase appears to appropriately follow the SPL when considering the disruptive impact of modes and reflections. I average these 5 speakers to better see the trend of the phase. I just ignore all the various midrange phase ripples. I sometimes confirm that all is well by turning off the SWs and measuring one main at 1m or so with the rePhase filter applied. I normally get a very smooth SPL and pretty linear phase from 80-20k when I do this.

Are you saying there is a better way to deal with this? I may well be missing some opportunities with my process so any comments you have would be appreciated. Below are some charts just to show what I am looking at prior to creating the phase correction filter that I use in FooBar.

REW SPL.png

REW Phase.png

Holm Avg of 5.png
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Old 20th February 2015, 11:13 AM   #626
jtalden is offline jtalden  United States
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One more to show Phase Vs Excess Phase.

REW Excess Phase.png
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Old 20th February 2015, 05:18 PM   #627
jtalden is offline jtalden  United States
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Pos, I understand now your comment regarding too much of the LF phase correction. The sample I posted had an extra phase rotation below 20Hz. I must have gotten confused in the rush to post an example. I don't normally make that mistake. Sorry for the confusion!
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Old 23rd February 2015, 10:02 AM   #628
micfre is offline micfre  Canada
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Originally Posted by micfre View Post
I worked up something in Excel, but I am not super confident I got it right! (zip attached)
Reload the same file in a working zip this time.

I attempted to work the math back from a paper. I may be close .. I may be way off. Please don't use this to design a filter until someone with experience chimes in!
Attached Files
File Type: zip HK critical frequency calculation.zip (7.4 KB, 19 views)
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Old 21st March 2015, 03:56 PM   #629
Laotzu1 is offline Laotzu1  Denmark
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I have been using rePhase for almost 2 years. It is very easy to use and gives excellent results and sound quality. The idea of focusing on manual corrections is very good, I like to stay in control and selectively choose corrections. Amazingly rePhase is free, but the value is in my opinion very high. So Pos, when are you ready to receive donations? For sure, I am ready to donate.

One wish for improvement: it would be nice with a short explanation for each field in the settings. Fx. in the impulse settings, centering: what are the implications of choosing “use exact centering value” and the other options?

My current and most advanced use so far is for a 3 way active speaker system. My use is somewhat similar to Pano’s. The filters are implemented in a PC feeding a multichannel DAC that directly sends the signal to power amplifiers. The procedure is: measurement with Holm, import to rePhase for amplitude and phase linearization. After this I am free to play around with XO settings for filter generation and import the filters to JRiver via convolver interface. So far I have primarily used LR 24 dB for XO. Results are very good with a lot of details, dynamics and a very well defined soundscape. Taps are the largest available, 65536.

I have formerly regarded brickwall filters as a primitive, brutal concept for XO, but based on post 557, I tried it. The windows were hamming and blackmann in accordance with textbook recommendations. Surprise. Imaging became even better, the sound was even clearer and fuller and sweet spot widened both horizontally and vertically. This is probably due to the absence of crossover regions where one frequency is played back through two loudspeaker units (reduced comb filtering and remaining phase differences?). The sound was not fatiguing or unpleasant in any way. I had expected some audible ringing, but nothing of the kind manifested. The only drawback is a low whistling tone setting in as soon as playback is started, which is about 2/3 second before music starts to play (FIR filter delay). What might be the cause of this whistling and does anyone have suggestions to get rid of it? The processing time for generation is a lot faster for brickwall filters compared to LR, is the generation/calculation in rePhase different, and can this the reason for different sonic quality? Does anyone have suggestions for improving use of brickwall filters for XO?

I have just read the entire 600+ posts in this thread. This calls for a few tips of mine regarding issues raised.

Measurement of delay between loudspeaker units in active systems is very easy using Holm. Just send the frequency sweep in parallel to two amplifiers and measure in listening position. The impulse screen in Holm will show two peaks, and time delay between peaks can be read on the horizontal axis after setting it to show to time instead of samples which is default. This is the delay to implement in JRiver or VSTconvolver.

DSP processing of sound from all windows applications can be achieved easily through JRiver version 20 because it comes with a WDM driver. By choosing this as Windows standard sound unit, sound is automatically processed through JRivers DSP and filters. No need for starting loopback as in previous editions.

Synchronization of sound and video is automatically optimized when playing movies through JRiver due to the “lip sync” function. Synchronization of sound and video from other programs can be achieved by setting a manual delay for video in JRiver.

My last remarks has developed into something similar to a commercial for JRiver – sorry about that – I am not paid by JRiver. It is just so easy to integrate all functions related to DSP filters in JRivers standard functionality, so I have to praise it for that. I am not so pleased with its user interface, but that’s another story.

Phuu, this is a long post. Once again, thanks, thanks, thanks for rePhase.
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Old 22nd March 2015, 02:09 PM   #630
pos is offline pos  Europe
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Hi Laotzu1

Quote:
Originally Posted by Laotzu1 View Post
I have been using rePhase for almost 2 years. It is very easy to use and gives excellent results and sound quality. The idea of focusing on manual corrections is very good, I like to stay in control and selectively choose corrections. Amazingly rePhase is free, but the value is in my opinion very high. So Pos, when are you ready to receive donations? For sure, I am ready to donate.
Thank you
I am pondering adding a paypal donation button when version 1.0.0 is released, but I am not sure yet.

Quote:
One wish for improvement: it would be nice with a short explanation for each field in the settings. Fx. in the impulse settings, centering: what are the implications of choosing “use exact centering value” and the other options?
You mean like an integrated help functionality?
Maybe a question mark button on the top corner of each tab?...
That is a good suggestion that I will try to implement for the 1.0.0 release

Regarding centering option, short answer is always use the ideal centering.
Other options might give you ripples around the impulse peak (that can be observed by loading the IR into holm) when the final fractional sample is different than the ideal value which artificially enforce a 0° phase as the Nyquist frequency.

Quote:
I have formerly regarded brickwall filters as a primitive, brutal concept for XO, but based on post 557, I tried it. The windows were hamming and blackmann in accordance with textbook recommendations. Surprise. Imaging became even better, the sound was even clearer and fuller and sweet spot widened both horizontally and vertically. This is probably due to the absence of crossover regions where one frequency is played back through two loudspeaker units (reduced comb filtering and remaining phase differences?). The sound was not fatiguing or unpleasant in any way. I had expected some audible ringing, but nothing of the kind manifested. The only drawback is a low whistling tone setting in as soon as playback is started, which is about 2/3 second before music starts to play (FIR filter delay). What might be the cause of this whistling and does anyone have suggestions to get rid of it? The processing time for generation is a lot faster for brickwall filters compared to LR, is the generation/calculation in rePhase different, and can this the reason for different sonic quality? Does anyone have suggestions for improving use of brickwall filters for XO?
Brickwall filers are indeed the most prehistorical form of FIR filtering, and very easy to do (windowed sinc...).
You must use the same window, sampling freq and number of taps for both sides of the crossover or the summation will not be perfect.
Using additional corrections might also affect the final complementarity...
If you want steep slopes it is probably better to use LR with high slope values and make sure your window will let the resulting curve track the target ones close enough.
Also try reject high of low crossover types if you want to reject specific frequency ranges from your drivers...

Regarding the noise, I have no idea where it can come from.
Try loading the impulse in audacity or holm and look for any weird thing...?
Jriver most probably implements its convolution in the frequency domain, which can also imply all sorts of warts when not done properly, although I am sure their implementation is just fine...
Maybe try a direct time domain convolution engine just to make sure?

Regarding brickwall being faster to generate, it probably comes from the fact that iterative optimizations is bypassed (same as "none" setting) when they are used, becuase it somewhat goes against the very principle of that kind of filter...
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