# New function required : implementation of Low pass and High pass filters to perfectly simulate response curve with EQs



## Philippe75

Hi everyone,

I am using REW for several months now, and I am slowly improving my measurement and optimization method. Quite time consuming, but when we like it… :innocent:

My subject of interest is a 3 voices multi amplified system with a Xilica crossover. I measured each speaker individually on an appropriate frequency range, near field method for Low voice and 1 meter distance for Medium and Tweeter voices. Once measures are OK, I ran several EQ wizards on each speaker curve to obtain an acceptable linear response curve using limited EQ gain values. Each speaker is also affected by High pass and Low pass filters implemented in the crossover (model is JMLC “almost perfect filter” based on 3rd order Butterworth filter). Therefore, I ran EQ wizard outside the -3dB filtering range to optimize the overall curve (i.e. within the filtering range, at crossing frequencies and outside).

The Generic equalizer can simulate High pass and Low pass filters. These filters are not implemented in the Xilica, so I have to swap between Generic and Xilica which is not perfect but acceptable. I found that I have to manually adjust EQ to better match the LP & HP filtered curve. However, I have no idea of what the characteristics of these LP and HP filters are (2nd order?) and no possibility to setup a filter of my own (Butterworth, Linkwitz–Riley or Bessel). No additional options in the Target Settings section… My manual EQs are maybe not that appropriate.

That’s why I suggest to create a new function that would enable to setup a Butterworth, Bessel or Linkwitz-Riley LP and HP filters. EQ would be defined as they are currently done, but simulating LP and HP filters would give an overview of the very final result. That would be perfect!!!

Hope to see this in the coming weeks…

Regards
Philippe


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## Philippe75

Hi everyone,

Is this request not a needed function by anyone else that me ?
Is there a way around to achieve this simulation that I haven't found yet ?

Thanks for your help and the overall quality of the software !

Regards
Philippe


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## jiiteepee

Philippe75 said:


> ...
> However, I have no idea of what the characteristics of these LP and HP filters are (2nd order?)
> ...


http://www.electronics-tutorials.ws/filter/second-order-filters.html


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## Philippe75

Thanks for the link, I now know what are the specifications of the filters. Unfortunately, these are not 3rd order Butterworth, so I can not simulate the digital filters applied to each voice to then optimize SPL curve with PEQ.

As I mentioned, but however my method is probably incorrect, my objective is to record each voice independantly (done), to identify optimal frequency cutoffs (done) for each voice thanks to manufacturers data and REW measures (done, but never good enough ), to apply filters to each voice and to optimize the curve.

I am applying this method because my system is 3 voices multi amplified, and simulations prevent me from doing live testing all the time. Any better idea to achieve my objective ?

Regards,


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## jtalden

Philippe75,
I am a little unclear on your objective, your current approach and any imposed constraints your may have, but will make the following comments in case they are helpful.

If the objective is just to find the best settings for a 3-way speaker using an Xilica speaker management box then I am be able to offer an alternative approach. We can take the approach that what is important is the phase tracking through the 2 XO ranges. 

If 2 drivers (voices) phase track closely it does not cause erratic SPL behavior in that range. The worse the phase tracking the greater the SPL is impacted both on and off axis. 

Driver phase tracking alignment is measured on the acoustic XO not on the electrical XO. It is not important if the Acoustic XO follows a particular textbook path like for instance a LR-24, or but-6, or JMLC “almost perfect filter”. If the drivers acoustical phase tracks well then any combination of electrical XO settings is optimized. That may be achieved for example with a But-24 LPF at 2k with a BES-12 HPF at 2.5k and 0.036ms Tweeter (TW) delay between the 2 drivers. In practice we can usually find several combinations of setting that meet this criteria when using a speaker management box like the Xilica. 

The JMLC “almost perfect filter” was designed to minimize the phase tracking/SPL issue given the constraints of a passive XO implementation. In that case there is very limited ability to "adjust" the delay. It is "locked-in" by the filter chosen and the driver offset. Active XOs are much more flexible as the delay can be set at any value.

Once the XO settings are determined via the filters and delays chosen then the final level settings and EQ can be done. If the EQ is done on the input of each channel the phase of all 3 drivers are impacted in the same way and the good phase tracking that was achieved in the XO ranges is not changed.


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## Philippe75

Thanks for your feedback, but I think we are talking of 2 different subjects. Even though, they are highly related.

In regards to JMLC filter, Jean-Marie Le Cleah managed to design a XO filter that focused on a flat SPL response at XO frequency, optimized group delay propagation at XO frequency and limited impact on phase tracking. This is the reason why it is called "almost perfect" as those 3 criteria are to be taken into account for best sound quality. To achieve this objective, the filter requires to respect physical positions of drivers using a given formula at XO frequency. This filter is easy to implement in a Xilica box as you mentioned. Not perfect anyway, but the best filter I tested so far...

From my understanding and readings, there are no perfect 12db or more XO as they all impact these 3 criteria, except maybe FIR filters... If you have any suggestion, I would be grateful to read from you if I can implement it in the Xilica Box. In regards to your comments, I haven't yet measured the whole XO result (3 voices) as I was still focusing on optimizing XO points and individual speakers' SPL curve. But from your comment, I understand that the SPL curve might be impacted by unpleasant phase tracking.

Below is a detailed presentation of my approach:
1) I measured each voice individually on the whole range of frequency (10 Hz - 22 500 Hz) respecting the signal path. I.e. PC -> Preamp -> Xilica (flat) -> voice Amp.
2) Based on measures and manufacturer information, I define ideal XO frequencies taking into account SPL curve flatness (no EQ), distortions levels...
3) Once XO are defined (long and iterative approach, hum), I simulate XO (JMLC is based on Butterworth 18 dB (3rd order) => *this is the missing function in REW, my request*.
4) I optimize resulting curve with EQ to get the best result, but I try to limit the effect of an EQ to where it is needed.
------ This is where I am
5) I will have to optimize physical position of drivers to have aligned impulse from the 3 drivers.
6) I will have to measure the whole result and see if the SPL curve remains flat at XO frequencies and if phase is not too erratic. I must admit this is not the criteria I understand the best…….

Any suggestions?


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## jtalden

There is nothing wrong with pursuing your path. I have tried all sorts of XO/EQ setup to decide for myself what makes a significant difference. 

I was just pointing out that the JMLC filter is designed within the constraints of a passive XO network. The active XO setup with the Xilica box removes those constraints. That would allow a more perfect XO to be created. It also allows several different types of perfect XO to be created for comparisons.

I think I now understand your question however. You would like REW to create a But-18 House Curve for you. In the meantime you are asking how to import a But-18 target curve into REW so that you can try to EQ the "Voice" to match it.

Any House Curve can be imported into REW. The REW "Help" explains the method. The part that is probably the issue for you is the creation of the data to import. 
1. Possibly someone will create the file for you to import. I did a spreadsheet for But-12 and LR-24. but it does not include But-18. There is no doubt a program available somewhere. I know it can be done easily in Matlab and probably many others so someone could provide you a trace, or better yet, the coordinates.
2. If you find a graph of the But-18 response you are wanting (possibly at the discussion site of the JMLC filter or in his presentation?) there is a program that will allow the coordinate file to be created: 
SPL Trace


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## Philippe75

From my understanding, passive or active filters have the same effects on SPL, group delay and so on. Am I wrong ? If this is not correct, active can get ride off passive drawbacks, then indeed I should test different XO structure.

In regards to room curve, I think this would be a too hard task because I need as much house curve as I have speakers and XO points... I don't think this would be a good approach. I still think the best would be to implement, in addition of the existing HP and LP filters, 2 parameters : filter type (Link, Butt, Cheb...) and order (6dB, 12 dB, 18 dB...). This being done, we could all optimize PEQ to the final curve. Not a too big deal, isn't it ?

I now plan to measure all speakers with XO and PEQ to see if time overall volume is linear and if adjustment is OK as you described.


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## jtalden

Philippe75 said:


> From my understanding, passive or active filters have the same effects on SPL, group delay and so on. Am I wrong ?


This is basically correct in one sense, but it is an over simplification.



> If this is not correct, active can get ride off passive drawbacks, then indeed I should test different XO structure.


The active box can create the classic electrical filters listed in its documentation accurately. A passive design is typically more limiting. They can approach the classical electrical filters for the low order (1st and 2nd order) reasonably well. Their accuracy is not as good because of; component tolerances, and the speaker impedance is often not uniform in the XO range. This make the higher orders problematic to implement. Regardless, the electrical XO accuracy is not the important characteristic anyway. It is the acoustic XO result that is important not the electrical settings that were used to get there. 

The biggest difference is the active filter will allow a delay to be set to align the phase tracking through the XO. This is one of the important consideration in XO design. With a passive design this design consideration is not possible unless the horn is moved forward such that the compression driver is much closer to the woofer than the normal situation with the horn mouth at/near the baffle. 

The JMLC design is for a passive XO and does not align the Phase tracking, hence the "“almost perfect filter” designation. It allows the phase to cross at the XO frequency at a high slope and thus the SPL is erratic in that range.

See the chart in *this post* for an example of a non aligned horn setup. The chart is windowed to remove the room effects so we can easily see the effect on the direct signal, oops, I mean the effect on the "Direct Sound".

There is nothing "wrong" with this type of XO alignment. It is common for all large horn speakers using passive XO filters and with the horn mouth at the baffle. I was only pointing out that it is not necessary to make this XO compromise when using an active XO box where delays can be set.



> In regards to room curve, I think this would be a too hard task because I need as much house curve as I have speakers and XO points... I don't think this would be a good approach. I still think the best would be to implement, in addition of the existing HP and LP filters, 2 parameters : filter type (Link, Butt, Cheb...) and order (6dB, 12 dB, 18 dB...). This being done, we could all optimize PEQ to the final curve. Not a too big deal, isn't it ?


I thought you wanted a target curve in REW that allows you adjust the Xilica XO filters and EQ to match that target. REW does not currently do that for us but allows us to input any target we want in the EQ panel. REW calls it a "House Curve", but it can be used to set any target curve we like, for any reason including, e.g., But-18 HPF at 1kHz. The horn voice can then be adjusted to match it. The target curve for the LPF can then be entered and the midrange voice adjusted to match that, etc. Once all the filters/voices are set then the overall house curve can be entered and the room EQ applied. Is that what you want to do? There are many DIY's that do it that way (even if I personally don't think it is efficient or advantageous).



> I now plan to measure all speakers with XO and PEQ to see if time overall volume is linear and if adjustment is OK as you described.


The XO alignment / phase tracking cannot be easily evaluated with an overall measurement. This must be done by measuring the voices independently with "loopback timing" turned on in REW. *This thread* has more info on what is involved in case you ever decide to try it that way.

That method is likely more complicated that you are interested in now, so you may just want to start with something easier. Just select LR-24 XO filters and adjust the delays using the REW RTA feature until the SPL is as smooth as possible in the XO range. That "easy" method will provide a similar result to the JMLC passive design.


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## Philippe75

I took some times to read carefully your post and links. There are very interesting and I think I was not clear enough in my first posts (my English is not that good)...

In regard to passive and active XO, you are perfectly right, time alignment is one of the key for a good signal response. My system is made of three speakers, and the medium is a horn which is "far behind" the woofer and the tweeter (40 cm). At the moment, I did a measurement of each individual speaker positions to align them all using delays. I still need to confirm measurements with REW as physical measurements are not precise enough... These delays are implemented in all XO I parameterized in the Xilica.


For the sake of the discussion, I don't understand your point when you say JMLC filter was designed for passive XO. Jean-Marie Le Cleah spent a lot of time seeking for the perfect XO that met all these criteria : SPL, minimum effect on group delay and phase rotation. His final proposal is (the tweeter is the reference) :
- LP and HP are Butterworth of the 3rd order, XO is designed at -5 dB
- LP XO frequency = 0,8729 * XO frequency (to match -5 dB)
- HP XO frequency = 1,1456 * XO frequency (to match -5 dB)
- Medium of Woofer are to be moved ahead = 0,22 * wave length at XO
- Medium phase is to be inversed
The result is a quite linear response curve, group delay and phase rotation is in between -150 and -20 degree.

I am not sayin it is the best as there are no best XO except maybe FIR, but it is a good approach to manage all criteria. The filter you are using in your link looks like Samuel Harsch proposal : LP=Butterworth 4th order, HP=Bessel of the 2nd order and a constant delay is to be set for HP=(1/fc)/0,5. I have not tested it so far, I will in coming weeks. For your information, JMLC also desiged a good Excel simulation sheet to test various kind of XO. you will find it on that link (French) : http://nicolas.davidenko.perso.sfr.fr/filtragejmlc/filtragejmmlc.html


In regards to REW new function required, I will test your proposal but I need to generate 3 house curve, one for each individual speaker. Just for your understanding, I found out that when I aligned one speaker curves using PEQ, some PEQ were not well adapted when applying the HP and LP. This is why I would like to simulate filters in REW to manually fine tune REW PEQ proposal directly.

Philippe


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## jtalden

Please understand that I recognize a JMLC recommended XO as a very good theoretical design. I would expect the sonic results to be excellent.

My understanding of the difference between JMLC approach and the alternate one I mentioned is that JMLC provides a analytical tool to predict the acoustic XO performance. It is a theoretical design. The alternate approach is to measure the actual acoustic XO response and adjust the Filters, Delays and EQ as needed to achieve the best measured results. 

If I understood correctly, your plan was to take a recommend JMLC XO design and implement that using the Xilica. I see no issue with that. 

It also sounded like you intended to try to EQ to the voices to match the theoretical/calculated response path of the individual voices. I do not see an advantage to that that approach. You could instead just implement the JMLC filters and delays; then fine tune the delays based on the actual measurements as needed, and then EQ to the overall response, i.e., the response of all 3 voices working together. I believe this approach negates the need import a But-18 target into REW. I probably should have ended my recommendation with this suggestion.

I went a little further and tried to point out that it is possible to just approach the whole problem using an empirical approach rather than analytic approach. That is, we can select the filters; measure and select the best delay for those filters (to achieve the best phase tracking), and then EQ to the overall response. If we are looking for optimized phase tracking, it may be necessary to change one or both filters in the XO to optimize the tracking. I have not studied the JMLC info to be sure, but understood that the phase tracking was not optimized in the particular design I saw. It appeared the phase of the voices crossed at the XO freq and diverged from each other in the rest of the XO range. This compromise was made to accommodate the limited offsets possible in a passive XO system. This may, or may not, be the case for the particular alignment you are considering.



Philippe75 said:


> For the sake of the discussion, I don't understand your point when you say JMLC filter was designed for passive XO. Jean-Marie Le Cleah spent a lot of time seeking for the perfect XO that met all these criteria : SPL, minimum effect on group delay and phase rotation. His final proposal is (the tweeter is the reference) :
> - LP and HP are Butterworth of the 3rd order, XO is designed at -5 dB
> - LP XO frequency = 0,8729 * XO frequency (to match -5 dB)
> - HP XO frequency = 1,1456 * XO frequency (to match -5 dB)
> - Medium of Woofer are to be moved ahead = 0,22 * wave length at XO
> - Medium phase is to be inversed
> The result is a quite linear response curve, group delay and phase rotation is in between -150 and -20 degree.


Possibly this particular alignment creates close phase tracking throughout the XO range, but the one I saw crossed at the XO freq as it was intended as the best solution for a passive XO. The example in the link your posted also may not be optimized for phase tracking. It is impossible for me to tell for sure as there is no phase tracking chart shown and I do not read French. It appears to be compromise design from what I can glean, but again, I am not sure.



> I am not sayin it is the best as there are no best XO except maybe FIR, but it is a good approach to manage all criteria. The filter you are using in your link looks like Samuel Harsch proposal : LP=Butterworth 4th order, HP=Bessel of the 2nd order and a constant delay is to be set for HP=(1/fc)/0,5. I have not tested it so far, I will in coming weeks. For your information, JMLC also desiged a good Excel simulation sheet to test various kind of XO. you will find it on that link


It is, no doubt, a good approach and since you are measuring, it will be easy to see just how ideal it is. 

The example in my link followed no theoretical approach or any recommendation. It was empirically determined to be a good solution for my particular setup of voices and listening axis. It was one good solution of several that I have found empirically.



> In regards to REW new function required, I will test your proposal but I need to generate 3 house curve, one for each individual speaker. Just for your understanding, I found out that when I aligned one speaker curves using PEQ, some PEQ were not well adapted when applying the HP and LP. This is why I would like to simulate filters in REW to manually fine tune REW PEQ proposal directly.


This is main point I was trying to help with. I was trying to provide you an alternate approaches that will provide the same or better result. One that allows you continue on with your setup with out the need to figure out how to enter these target curves into REW. After all, you did ask for options in Post 2.


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## Philippe75

Don't make me wrong : I highly appreciate your feedbacks, sharings, and very detailed explanations. :bigsmile:

I will test your approach once I will be back and able to install the measurement tools again. I will keep in touch in that thread to let you know in coming days.


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## Philippe75

Hi,

Following the method described in given Thread, I tried today to measure "raw" delays in between individual speakers, i.e. without any filters. I wanted to be trained with the method by measuring delays generated by the design of the speaker (the Supravox is roughly +40 cm in front of the Beyma and +8 cm in front of the Fostex). I activated the "Use Loopback as Timing Reference" setting in REW, but I get 3 impulses almost at 0. Same results if I add a 1000 mm delay in Xilica as proposed in thread...

Following is the my measurement setp :
- MiniDSP UMIK1 USB mic attached to a laptop
- USB link in between the laptop and the DAC of the PreAmp - the soundcard is not used.
No physical loopback... and I don't see how to design one.

Any idea on how to setup a loopback of force REW to not align impulses on 0 ?

Thanks in advance.

------
Up date : I just see 2 posts below which deal with this issue (dealing with UMIK). I will have to find a way with my desktop I think...


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## jtalden

Yes, the popular USB mics are not compatible with loopback timing. This makes it impossible to use them with the current version of REW to fine tune the phase handoff between voices/drivers. They are fine for all, or at least most all, other acoustic measurements. This is also a limitation if fine tuning of the delay/distance settings for a SW to main speaker XO is needed. However in the SW to main speaker case, it is reasonably easy to find good settings without using loopback timing. That's because the wavelength is so large at low freqs a ±0.5 m error in distance does not have major impact on SPL or sound quality. The midrange is not a forgiving. 

It is too bad that many of the DIY speaker builders using a speaker management box like your Xilica or my DCX do not realize this limitation when deciding on the mic to purchase. 

I expect it is possible to use a typical internal soundcard loopback with a typical "cheap computer mic" for this type of phase alignment work. Once the delays are established then the USB mic can be used for EQ and any other analysis that is needed.


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## Philippe75

Yesterday, I purchased a Taskam US-125M external USB sound card with 2 Line- out and 2 Line-in slots. I still have my old Ecm8000 mic, so I should be able to achieve the delay measure in coming days…


Taking a deep look at Ecm8000 and Umik-1 measures, I found-out that, if SPL are comparable, there are big discrepancies in GD, Impulse and Phase. Globally speaking, Umik-1 measures are weird compared to Ecm8000 ones. For instance, Supravox impulse starts normally with Ecm8000 but reversed with Umik-1; Beyma phase is weird below 600 Hz with Ecm8000 but still weird at 2K Hz with Umik-1; GD is almost flat with Ecm8000 but weird again with Umik-1 below 2K Hz.

To be complete, Umik-1 measures were done through a USB-to-SPDIF bridge connected to the Accuphase DAC-20 extension board when Ecm8000 measures were done through the PC sound card (calibrated) connected to one analog input of the Accuphase. I will have to make some measures with Umik-1 and the PC sound card to see which component is generating such results…

Any idea?


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## jtalden

I wouldn't expect any significant difference in the Phase/GD results of the 2 setups, but...

At the LP even a very minor change in the mic position can have a large apparent impact on the charts as the room reflections may be very different. With the mic at 1 m a small position difference should be negligible because the direct signal is much strong than the reflections for the mid to upper frequency ranges. The lower freqs can still be problematic.

With proper scaling, windowing and filtering of the measurements, we can often see the phase trend of the direct sound even with the mic at the LP. It is the direct sound phase that we need to accurately measure in order to properly align the XO handoff.

I would need to see the .mdat file and understand your test conditions fully to provide a response specific to your situation.


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## Philippe75

I found a mistake in my analysis yesterday: I forgot to adjust the IR Window... :R Once done (depending on measures spec), results are much better for Phase and GD but Phase remains very different even if curves have now similar shapes.

Following graphs are of the Beyma horn speaker. In GREEN, Ecm8000 measure. In BLUE and RED, 2 different Umik-1 measures done in January and May but with similar audio path and physical mic location. For the 3 measures, mic distance to the floor is 1 meter and mic distance to the horn speaker mouth is 1m for Umik-1 and 70 cm for Ecm8000 (old measure). The major difference (to me) is that Ecm8000 measure was done through the PC Soundcard and Preamp analog input whereas Umik-1 measures were done through the USB to SPDIF bridge and Preamp DAC input. Frequency range is 200 Hz to 22 050 Hz.

*SPL*








*Phase*








*Group delay*









However, Umik-1 impulses remain very different compared to Ecm8000. As if there was a 180° applied to the speaker which can also be seen in Phase graph.


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## jtalden

Philippe75 said:


> ...The major difference (to me) is that Ecm8000 measure was done through the PC Soundcard and Preamp analog input whereas Umik-1 measures were done through the USB to SPDIF bridge and Preamp DAC input.


Just to clarify/confirm: The Tascam/ECM setup should bypasses the internal PC soundcard. The Tascam output should go directly to the Preamp input.

Also:
You are correct that the polarity is reversed for one of the 2 measurement setups. Using the Tascam/ECM setup this is easy to determine and correct as needed during the Tascam loopback calibration process. Neither of these setup should result in inverted polarity however. If the internal PC soundcard was actually used then it would not be surprising if it inverts the polarity.

There are a couple of other minor observations that are of no concern at this point.
> The SPL calibration for the 2 mics is significantly different at high freqs. Possibly the ECM is not calibrated or the appropriate mic calibration files were not loaded. This has no impact on phase/GD charts.
> The red UMIK-1 phase at the high freqs has an issue. Without looking at the .mdat file it is impossible to tell if that is just related to the REW settings applied or possibly to an issue with that particular measurement. Since we will not be using that particular measurement, it is irrelevant.
> REW allows the position of the IR to be offset as needed to align with other measurements. We could invert the ECM IR to match the polarity of the two UMIK-1 measurements and then offset the 3 IRs as needed to line them up. That way the phase chart overlay would track very similarly. They would overlap until the red trace falls away a 12kHz. This is just a convenient way to better depict how closely they track.

The Tascam/ECM setup is ideal for the process of adjusting the delays and XO setting to achieve close phase tracking throughout the XO range.


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## Philippe75

Yes, you are right. When using the Tascam card, I will have to generate a calibration file and then use the board directly connected to the Pre and Ecm8000 (back to analog measure). I will not use the laptop internal soundcard.

In regards to mic, you are also right : the Ecm8000 is using the generic calibration file proposed with REW as it was not calibrated when I purchased it. This probably explains the SPL curve at high frequencies. To be honnest, I had no idea of calibration when I purchased it 3 years ago... This is one of the reason why I purchased the Umik-1 mic early this year (in addition to the simpler USB interface but with limitation).

As far as polarity is concerned, the GREEN curve was measured with the internal PC sound card and the Ecm8000. I have no idea if polarity was reversed by the card or if the measure was done using a specific parameter set by error. I can e-mail you the .mdat file if you don't mind.

Regarding high frequencies SPL measure, I have this issue at 13 200 Hz with both mics. I thought this could be an effect of the horn... I will anyway set a LowPass filter at 9 000Hz max. You may find some information about the horn on this french site : http://www.guigue-locca.com/pavillons.html - reference is 300 C1.4.


By the way, one question : as you can see, the Beyma is naturally having a strong curve below 800 Hz. I always wondered if it was good to setup a HighPass filter which will reinforce the curve or let it as it is with a HP set low (at 300 Hz for instance) to protect the speaker...

In regards to the topic initial subject, I recently found that I can generate a HP or LP (Link) impulse using rePhase and import it as a .wav file to simulate the effect of the filter on a SPL curve. I just have to generate the different HP and LP required and use the A*B calculation provided within REW. Simple solution for RAW curve, but I have to find how to save an parametric equalized curve first as I can not apply the function on a target SPL curve...


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## jtalden

Philippe75 said:


> As far as polarity is concerned, the GREEN curve was measured with the internal PC sound card and the Ecm8000. I have no idea if polarity was reversed by the card or if the measure was done using a specific parameter set by error. I can e-mail you the .mdat file if you don't mind.


You can post .mdat files on here. No need for e-mail. There is no need to post this file though as there is no open question that I am aware of. It is not unusual for an internal soundcard to invert the signal. The Tascam will not do that. I don't see that these old measurements need any more analysis as they cannot be used. 



> Regarding high frequencies SPL measure, I have this issue at 13 200 Hz with both mics. I thought this could be an effect of the horn... I will anyway set a LowPass filter at 9 000Hz max. You may find some information about the horn on this french site : http://www.guigue-locca.com/pavillons.html - reference is 300 C1.4.


That is a beautiful horn!



> By the way, one question : as you can see, the Beyma is naturally having a strong curve below 800 Hz. I always wondered if it was good to setup a HighPass filter which will reinforce the curve or let it as it is with a HP set low (at 300 Hz for instance) to protect the speaker...


It is normally recommended to set the XO an octave or more above where the SPL response starts rolling off. It looks like the manufacturer suggests 400Hz as a minimum useful lower limit. If you are intending a 3rd or higher order filter then any freq higher than that will be okay from a safety perspective. I would have expected the best XO to be higher than that however. Normally the lower limit is a function of the horn mouth size. Below that starts to cause issues, but I am not really skilled in best horn practices. It does take a pretty large horn to reach 400 Hz comfortably. 

A lot also depends on the capability of the lower voice. It's best to stay in the comfort range of both voices. You can experiment with various XO points if you like or just pick one that you think is a good compromise. I tend to try to split the difference in the overlap capabilities of the 2 drivers. A good 15" will go to 400-500 and a good 12" to 800 or so.



> In regards to the topic initial subject, I recently found that I can generate a HP or LP (Link) impulse using rePhase and import it as a .wav file to simulate the effect of the filter on a SPL curve. I just have to generate the different HP and LP required and use the A*B calculation provided within REW.


I hadn't thought about using RePhase that way - interesting.



> Simple solution for RAW curve, but I have to find how to save an parametric equalized curve first as I can not apply the function on a target SPL curve...


I suppose there are programs that allows IIR filter settings to be input and will then create and save the IR, but I am not aware of them as I have not had the need for it.


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## jtalden

It just occurred to me that your above question regarding the 300 HPF filter is maybe in reference to a process where the EQ is applied to make the voice flat before applying the XO. If you do that process then keep the volume low and set the REW sweep to start at 300Hz or your idea of what is safe and required. That can be done in the measurement pop-up box. 

I would avoid any process that results in any significant EQ boost in the cut off region of a voice due to both safety and distortion concerns for the voice.


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## Philippe75

No, my point was about the "natural" slope of the speaker below 700 Hz which is close to a 18db HP filter. I was wondering if there was an interest in not implementing a HPF to take the benefit of this natural slope. But I was also keeping in mind that there should be a problem passing too low frequencies in the speaker. So a solution could be to implement a lower HPF to protect the speaker only. But I don't know if this makes sense...

In regards to PEQ, I am only willing to use PEQ to linear the curve in between HPF and LPF XO points. Not above or below. But based on all your feedback and learning a of the past days, I don't know if I should spend too much time on this since phase and then SPL is the basis.

So far, thanks again for all informations shared with me !!!


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## Philippe75

You will find attached one Umik-1 measurment of the Beyma speaker (raw measurement).


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## jtalden

Philippe75 said:


> No, my point was about the "natural" slope of the speaker below 700 Hz which is close to a 18db HP filter. I was wondering if there was an interest in not implementing a HPF to take the benefit of this natural slope. But I was also keeping in mind that there should be a problem passing too low frequencies in the speaker. So a solution could be to implement a lower HPF to protect the speaker only. But I don't know if this makes sense...


Then my original comments regarding XO selection are appropriate (post 20).


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## jtalden

Philippe75 said:


> You will find attached one Umik-1 measurment of the Beyma speaker (raw measurement).


Interesting.

I am confident the REW phase chart is correct, but, for me, it has an inflection at about the 3kHz range. I believe it is the real characteristic of this particular horn and driver. I just haven’t seen one measure like this before. Possibly it is common as I have only worked with a limited number of horns.

Other comments:
> The polarity appears to be negative as the initial rise of the IR is negative. The UNIK-1 and REW is not the source of the inversion so it must be elsewhere. Very possibly the unusual phase response of the horn causes this effect? This is not really an issue though because the tweeter (TW) polarity is normally set to a positive polarity. The midrange (MR) polarity and delay are selected to provide the closest phase tracking with the TW. Then the woofer (W) polarity and delay are selected to provide the closest phase tracking with the MR. It is not required to start with the TW as positive, but it is convention and it helps a little in order to simplify the analysis. The polarity of the MR and W that create the best phase tracking throughout the XO ranges will be impacted by the filter slopes chosen and the driver characteristics so there is no way to simply state which polarity is “correct”. If the acoustic phase slopes are very near LR-24 then of course all the drivers will be the same polarity. If we select XO settings to best accommodate the 2 voices natural characteristics then it is good to be open minded and select whichever settings provide the best measured results.

> My frame of reference in the bullet point above is with my process of setting up an XO empirically in mind. If you want me to help work through an example of that process using your measurements, I can do that. Just advise me that this is your intent and I will provide the first steps. If you instead intend to use the initial process you mentioned, or some other one, then let me know that also. That way I will try to avoid confusing you with more comments like this that do not pertain to your approach.


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## Philippe75

All measures of the Beyma speaker + horn give the same result at 1m. As you mentioned, I suppose this is the normal behavior of this association as manufacturers measures are not equivalent (but usually, manufacturers measures always look better, no?).

Based on your findings, REW and Umik-1 are OK. Good! So, I suppose that the inverted phase may be caused by the USB Bridge -> Accuphase internal DAC audio path. I will do a new batch of measures once the Tascam audio card is there and calibrated. The audio path will be Tascam -> Accuphase analog input -> Xilica -> Amp -> Umik-1. I then will be able to see if phase is back to normal or no. It was when I measured with audio path : Internal audio card -> Accuphase analog input -> Xilica -> Amp -> Ecm8000.

I would be very happy if you could drive me in this attempt. Don’t worry, I am very open-minded! I just want to understand what is happening as most of my “limited” knowledge is based on readings and theory. Thanks very much in advance!


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## natehansen66

I'd like to add a vote for the request in this thread (if I understand it right) that it would be huge for a xo designer to be able to generate target xo filters in REW. I've done the loopback and import house curve method in the past but it can be tedious if you are trying out different filter topologies, corner freqs, etc. ARTA has this ability and it's my go to in these situations, otherwise I usually prefer REW.

In my case I use JRiver for xo and eq. It only has the ability to do Butterworth filters (and LR types with cascaded 2nd order BW) so if I want to try different topologies I'm SOL without knowing what they actually look like.


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## jtalden

Philippe75 said:


> All measures of the Beyma speaker + horn give the same result at 1m. As you mentioned, I suppose this is the normal behavior of this association as manufacturers measures are not equivalent (but usually, manufacturers measures always look better, no?).






> I would be very happy if you could drive me in this attempt. Don’t worry, I am very open-minded! I just want to understand what is happening as most of my “limited” knowledge is based on readings and theory. Thanks very much in advance!


Good.  I think you will find this process results in closer phase tracking and SPL flatness than can be obtained using analytical process alone. While the empirical method it is complicated and confusing at first, I find it to be much easier to learn and implement than the analytical approach. 

> I suggest we first do the MR-TW XO together as it is easier because there are no room modes to muddle the charts. That will establish the process. Then maybe you will want to try the W-MR XO yourself. I will still help as needed.

> Let me know when the loopback calibration of the Tascam setup is completed and you are ready to start measuring voices. I will then provide setup/test instructions.

> It will be helpful if you provide (for example): W and TW ID/Specs?, Box type for the W (vented, or ??)?, Amps used?, Room Size?, Speaker and LP locations, Speakers toed in? , Music system only?, Music server in use? 
This basic type of info will help me stay on course and avoid problems stemming from assumptions about what your setup is.


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## Philippe75

To answer to natehansen66, yes my initial request was to introduce XO filters simulation in REW so that we can simulate impacts on SPL, phase, etc. I still think this would be a great improvement for esigning and testing XO.




> > Let me know when the loopback calibration of the Tascam setup is completed and you are ready to start measuring voices. I will then provide setup/test instructions.


Unfortunately, the Tascam US-125M is not the good external card... It is a mixer that combines multiple inputs (line, mic and instrument) and sends back the mixed signal to line output or the computer. Therefore, I cannot monitor the mic individually as it is mixed with REW SPL. In result, I get a loud interesting larsen... Nice ! I will change for the US-122M or US-322. If you have any idea before I send it back for an exchange...




> > It will be helpful if you provide (for example): W and TW ID/Specs?, Box type for the W (vented, or ??)?, Amps used?, Room Size?, Speaker and LP locations, Speakers toed in? , Music system only?, Music server in use?


*Amps*
- Accuphase E350 + DAC20: used as a PRE and as an external AMP for the Beyma voice (middle)
- Microméga PW400: used for the Supravox voice (low)
- Myriad Z62: used for the Fostex voice (tweet)

*Digital crossover*
- Xilica XP3060 : audiopath is Accuphase -> Xilica => 3 amps

*Speaker*
- The system is based on the Acanthe speaker you can find in the link "Guigue & Locca". It is a vented box for low voice + horn for the middle + tweeter. All voices are independent. The overall level is close to 98 dB.
- Woofer: Supravox 285 GMF
- Middle: Beyma CP755Nd + Guigue Horn 300 C1.4
- Tweeter: Fostex T90A

*Room*
Room dimension: 8,04m * 5,53 m, height is standard 2,5m (appartment).

*Usage*
The system is for music mainly, I also linked the TV box on it but it is for convinience. Sources are a laptop with an old version of JRiver Media Center and a CD player. The laptop is linked to the Accuphase DAC using a USB to SPDIF bridge (DAC20 option card is only SPDI/F).

Here it is, I have to wait for the new external sound card now.
Keep in touch.


----- Update
I anyway did some measures using the Tascam and Umik-1. Phase is still inversed for all speakers on impulse. I was anable to test back with Ecm8000 because of the Mixer issue with the Tascam...


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## jtalden

Philippe75 said:


> To answer to natehansen66, yes my initial request was to introduce XO filters simulation in REW so that we can simulate impacts on SPL, phase, etc. I still think this would be a great improvement for esigning and testing XO.


There are several here at HTS interested in this feature and maybe JohnM will implement it someday. For anyone that needs a workaround, I just confirmed Philippe75's comment that RePhase can create a wide variety of filter shapes that can be converted for use in REW. It takes a few steps to do, but it is not too difficult. See the chart below as an example. I created 200Hz Low-cut 18 dB/octave and 6kHz High-cut 24 dB/octave filters in RePhase. I then converted the file for use as a target curve (house curve) in REW. I then loaded a random full range measurement and used the REW EQ window to calc EQ filters from 150-7k Hz. It worked fine. RePhase can create any freq, any slope rate, and can even stack filters so this is a good workaround if needed.










> Unfortunately, the Tascam US-125M is not the good external card... It is a mixer that combines multiple inputs (line, mic and instrument) and sends back the mixed signal to line output or the computer. Therefore, I cannot monitor the mic individually as it is mixed with REW SPL. In result, I get a loud interesting larsen... Nice ! I will change for the US-122M or US-322. If you have any idea before I send it back for an exchange...


Do your have the Tascam "Loop Mix" switch set in the "Off" position? If it is set "On" feedback will be created. All of these USB audio interfaces have a feature like this using various names that will create this problem for REW. This interface will work as well as any other if the settings of the computer and the USB interface are appropriate. Someone here will be able to help you troubleshoot any problem you have.

Thanks for the info on your system. I will look up your voices and get my idea for XO points. Let me know if you have particular XO idea for the MR-TW. We just need a freq and the filter slopes as a starting point. We may later decide to adjust these if that is needed to achieve good phase tracking.


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## Philippe75

Just received the new Tascam box, I should now be able to achieve testing in good conditions. But unfortunately not before next Wednesday as I will not have the ability to save time for installing and testing the box before that date.

I was working on a MR-TW XO at 8 kHz or a little higher. Indeed, the Beyma speaker (if the measure was correct) becomes slightly erratic above 8 kHz and more beyond 10 kHz, and Fostex recommended XO frequency is 7 kHz or higher. In addition, I wanted to rely as much as possible on the Beyma to limit XO effects on key frequencies, and ideally respect the MR-TW XO > 10 * SW-MR XO rule (theory again).

Any advice so far?


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## jtalden

8kHz sounds fine to me.

I took a quick look at the published responses of the 3 units. There appears to be significant overlap available so there is good flexibility in the XO point selections and the filter orders for both XOs.

I would suspect that the MR starts to narrow its high freq dispersion rapidly above 9.6k (depends on the design of the throat of the horn) and would think the horizontal dispersion of the MR and TW may match a little better at less than 8k. I was thinking that 5k-6k may be a better guess for the XO. I couldn't find any horizontal dispersion info for the MR however so I really have no evidence to support this. Even if the best match does happen to be a bit lower than 8k I would not expect it to be a very significant sonic impact in practice. 

The Fostex 7kHz min XO recommendation has in mind a first order XO as shown on their schematic. They also have in mind the 50 W max power it's rated for. With such a high sensitivity I would doubt that you would even put 1 W to it. 106 dB above 5k Hz in a typical room is huge. If you are intending a higher order filter or not intending to push the limits of its output then it will be perfectly safe to XO as low as 5kHz in my opinion. 

My thoughts above are all just guesswork so let's do the 8kHz. That is as good a starting point as any. You can always try something lower or higher later if you like. At this XO you can use whatever filter slopes you like to start. As we review the measurements we may decide to adjust a filter slope slightly to achieve the best phase tracking. 

Whenever you are ready the initial 2 measurements needed are:
> TW (One channel, L or R)
> MR (same channel)

Conditions:
> If you have distance settings in the Pre-Pro just set them all to equal distance. We will not be using that feature.
> Set 8k Hz XO filters in the Xilica Use whatever filter slopes you want to start.
> Set 400 Hz? XO (Use your initial desired target freq and filter slopes) I would suggest a 12 to 24 dB/octave HPF for the MR if the XO freq is this low. If you go higher, to maybe 600, or more then a But-6 could be possible also. I'm just trying to avoid significant output from the horn below where it unloads.
> Set all Xilica delays at 0 ms.
> I suggest we start with no EQ activated. EQ can be left on for delay adjustments once we have it set reasonably.
> REW Loopback Timing engaged
> Full range sweeps, Approx, 20-20k Hz for all this work. We can run this 8k XO effort (2k-20k Hz) if you prefer. I usually run full range for everything as it has a couple minor advantages. Your choice. Just be sure all XO filters are set properly and active. This means the lower XO needs to be in place also and set to reasonable values. 
> Mic at LP and speaker in its normal position. The mic orientation is not critical for phase response, but the impact of reflections at high freq will be reduced slightly if the mic is pointed foreword (at or in the general direction of the speaker). 
> Approx 75 dB is okay for the measurement level.

I hope I remembered of all the important bits!

Post the .mdat file. I will review, and adjust polarity and delay for the best phase tracking. You can do the same if you followed and understood the link I posted earlier. I will then explain; what I did, where we stand, and recommendations for better tuning. If your just looking for good results rather than all the confusing detail, I will save effort and skip the explanations.


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## Philippe75

Done!

The new TASCAM is the good one. I also take the opportunity to buy a better mic with its calibration file (Audix Tm1 plus). All speakers have been measured with following audio path: Tascam -> Accuphase -> Xilica -> Voice amp -> TM1. Impulses look correct now.

I followed your instructions to measure Tweeter and MR with REW Loopback function activated. I can now clearly see delays in between TW and MR.

I have done several measures with different XO types and slopes (Link, Butt and Bess for 12 dB to 36 dB) to visualize effects of each filter on the phase. Following parameters are standard to all measures:
- MR: Highpass XO=500 Hz, Lowpass XO=8000 Hz (a starting point as you mentioned)
- Tweeter: Highpass XO=8000 Hz
- Mic: distance to the box=100 cm, pointing the middle of Woofer-Tweeter distance
- No delays
- Normal (positive) phase
- No PEQ at that step indeed

MDAT files (one for each XO type and slope) are huge, even zipped. I have to find a way to post one here. Tell me if you have a preference in terms of filter type and slope.

I have to analyses and understand results now. I would be very grateful if you could explain steps, findings and recommendations as I am interesting in understanding things.


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## Philippe75

Attached is the Link 24dB MR and TW measures file.


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## jtalden

Philippe75 said:


> The new TASCAM is the good one. I also take the opportunity to buy a better mic with its calibration file (Audix Tm1 plus). All speakers have been measured with following audio path: Tascam -> Accuphase -> Xilica -> Voice amp -> TM1. Impulses look correct now.


Looks good!



> I followed your instructions to measure Tweeter and MR with REW Loopback function activated. I can now clearly see delays in between TW and MR.


Yes, as we expected, the MR is significantly delayed as shown by the overlay of the 2 IRs. The approximate excess delay of the MR in this measurement is about 1.16 ms.



> I have done several measures with different XO types and slopes (Link, Butt and Bess for 12 dB to 36 dB) to visualize effects of each filter on the phase. Following parameters are standard to all measures:
> - MR: Highpass XO=500 Hz, Lowpass XO=8000 Hz (a starting point as you mentioned)
> - Tweeter: Highpass XO=8000 Hz
> - Mic: distance to the box=100 cm, pointing the middle of Woofer-Tweeter distance
> - No delays
> - Normal (positive) phase
> - No PEQ at that step indeed


All good, except mic position. I requested LP mic position and your intended speaker position. Maybe you are not testing with the room already completed or setup. This is no problem, but ideally the mic should be located on your intended "listening axis" we could also think of it as the line-of-sight axis. if we sit at the LP and look at a midpoint point between the MR and TW (vertically), that will be the listening axis. 

For example, If we intend to use a standard equilateral triangle setup and to face the speakers straight forward, then the listening axis is not directly in front of the speaker. it would be 30° off the speaker horizontally. If the mic was position within 20° or maybe 30° of the intended listening axis horizontally then that is probably an insignificant difference. We should be much more careful vertically. We want the forward lobe to be faced directly at the LP, not tilted up or down. That lobe will be very narrow so the sound will change significantly with vertical position changes, particularly so because of the large center-to-center distances of the voices and the high XO point. I would target being within ±3° of the intended vertical listening axis.

Also, 1 m is very close for a speaker this size. We should consider something nearer 2 m (if the LP is not convenient or available). This will help reduce error in locating the mic on the vertical listening axis.



> MDAT files (one for each XO type and slope) are huge, even zipped. I have to find a way to post one here. Tell me if you have a preference in terms of filter type and slope.


The slopes you provided appear to be pretty steep. I think that is a good idea in this case. It will to keep the XO range relatively small. The drivers are large and the XO is very high so this choice will minimize the overall disruption to the SPL response by limiting it to a narrow range. LR-24's are popular and have some benefits. Since we have wide overlap, I would have suggested we start there. Others may work out just as well so if you prefer to try LR-48 or something else that is okay.

Minor points:
> The REW SPL meter doesn't appear to be calibrated correctly unless it was extremely loud when the measurements were taken. I would suggest either setting it with an SLM or just calibrate the REW meter at 75 dB using a comfortably loud PinkPN signal. Testing should then be done to something near 75dB.
> It would be best to use the Xilica to decrease the MR level a little so it is closer to the TW level. We probably will want some HF decrease above 8k though so MR level can be a little higher than the TW. Maybe a MR decrease of 4 or 5 dB would be about right.
> Please set REW to measure using only 1 sweep for the measurements. There is no issue using 2 sweeps, but the file size is 2x as large with no benefit. Multiple sweeps are only important for certain types of measurements and phase timing is not one of them.

Path forward:
> I can use these measurements for timing alignment as we planned, but please advise me if you want me to do so. These are not suitable for best results unless the mic is at least close to the intended listening axis as defined above. Please advise.


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## Philippe75

Many points in your last reply. I try to answer in disorder.

2 sweeps : I read that this could avoid problems in "noisy" environments. As I am living close to a road, I thought this could help but indeed file size is huge...

SPL Meter calibration : I forgot to calibrate it indeed. I will have to do this next time. What could be the effect on measures ?

XO type : as mentioned, I measured several types of XO, i.e. Link at 12 dB, 18 dB, 24 dB, 30 dB and 36 dB. Same measures done with Bessel and Butterworth. I have not defined the XO Type yet, so tell me if you prefer another file to work with. On my side, I will have the opportunity to see differences in effects of XO to SPL and phase.

Regarding the mic position, I followed instructions given in the link Aligning Driver Phase. Too bad if it is too close or aligned with speakers but not the listening position. However, speakers are globally oriented towards the listening position (minus 10 deg probably). Is that really an issue ?

SPL level : yes, the 3 speakers have different levels. Current programs I defined in the Xilica take this into account usually -5 or -4.75 dB for Beyma compared to Fostex.


Well, I will not be able to perform new measures right now, so tell me if it is really blocking. I don't want you to waste your time on bad data...


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## jtalden

Philippe75 said:


> XO type : as mentioned, I measured several types of XO, i.e. Link at 12 dB, 18 dB, 24 dB, 30 dB and 36 dB. Same measures done with Bessel and Butterworth. I have not defined the XO Type yet, so tell me if you prefer another file to work with. On my side, I will have the opportunity to see differences in effects of XO to SPL and phase.


The XO filters chosen affect the job we are doing. We do not know in advance what will work best so that is why we measure. Your current XO settings (But-24? or LR-24?) is a good starting point for your speakers. The data provided is good in that respect.



> Regarding the mic position, I followed instructions given in the link Aligning Driver Phase. Too bad if it is too close or aligned with speakers but not the listening position. However, speakers are globally oriented towards the listening position (minus 10 deg probably). Is that really an issue ?


That's good for horizontal position. 10° horizontally is no significant issue as I previously mentioned. If the mic was the right height for the listening axis then all is well. I was a little concerned when you mentioned it was on the MR height. If it was same height as the throat of the MR horn at 1m distant then that is maybe 10-20° degrees below the listening axis. If we align the timing to a an axis that is 10 or 20° below the listening axis then the center lobe of the XO range sound will be pointed down at that angle instead of pointed at the LP. This will likely cause more instability in the vertical uniformity. The SPL may measure low and irregular in the XO range. This can affect the sound balance with small differences in vertical listening position. We normally want the sound field to be as stable and consistent as possible in the general area around the LP. Note that if we want to be able to confirm the same timing and SPL consistency at the LP as we get at 1m then any difference in the mic position vs the listening axis will impact the results. I take your comments here now to indicate you are satisfied with the positioning of the mic. That being the situation, I will now see what timing works best for this data. 

Some links:
Lobe info in general
Lobe - Example



> SPL level : yes, the 3 speakers have different levels. Current programs I defined in the Xilica take this into account usually -5 or -4.75 dB for Beyma compared to Fostex.


That won't prevent us from doing this analysis. It was just a minor point. I will just offset the MR data accordingly before doing the analysis. It was more a suggestion for any future measurements.



> Well, I will not be able to perform new measures right now, so tell me if it is really blocking. I don't want you to waste your time on bad data...


Only you can tell me if the data is good for moving forward. I outlined the preferred setup and my judgment on what amount of deviation is concerning from my perspective. I think I understand now that your only concern was with the 1m mic distance and not with the mic position vs the listening axis. The 1m mic distance is not a major concern for this work so we can move forward.

I just wanted to avoid spending significant time in analysis and results reporting to learn the initial settings were not appropriate to get good results. If I am successful in explaining the process then you can always make any needed adjustments yourself. In that respect, it doesn't matter. 

I will start working on data today.


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## JohnM

Philippe75 said:


> MDAT files (one for each XO type and slope) are huge, even zipped.


To keep down the mdat file sizes make sure the Analysis option in Impulse Response Calculation is set to "Truncate after 1.7s" and in Frequency Response Calculation tick the box to "Allow 96 PPO Log Spacing".


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## jtalden

The analysis is complete. 

Note: Both voices were first shifted in SPL level to place them near 85 dB primarily just for ease of analysis.

The best phase tracking is found by shifting the MR IR +5.825ms and the TW IR +4.698 ms. This is effectively adding a 1.137 ms delay to the TW so that its phase tracks with the MR through the XO range. The rest of the shift (4.698 ms) is just needed to mover both IRs to near 0 ms so that phase can be charted clearly. That allows fine tuning of the timing for closest phase tracking.

Results:
Below the A + B SPL is plotted for the original timing vs the shifted timing. Note that the original timing results in comb filtering of the direct sound SPL such that there are several significant dips in the response through the XO range. This charts also serves to show that the timing impacts the SPL from about 4k-17k Hz. That identifies the XO range of interest for this setup. Outside of this range there is no significant impact of the timing.









Below is the SPL of the XO region to better show the handoff of the XO for the shifted IRs.









Below is the original IR locations and the shifted locations:
















Below is the phase tracking that results from the shifted IRs. It also includes the A plus B trace showing the combined phase will track the MR before the XO point and then the TW after the XO point.









Below is the same phase tracking chart with several minor chart settings changed to help clean up the presentation of it. The IRs were; inverted, windowed appropriately, and the phase unwrapped to better show the phase tracking. Note that the tracking is very close from 6k-17k Hz and only departs starting at 6k until 4k Hz where the discrepancy is about 67° as shown. This is a very good result and the best that can be done with this setup. It may be possible to do a little better in the 4-6 kHz range by change the slope of the 500 Hz MR HPF to the next higher slope, but that change may not be effective and there would probably be no significant audible impact.









This completes the overview of the analysis. It's very difficult to provide step by step instruction for this type of analysis. It took me several years to do this relatively efficiently and it still takes me several hours. Instead of providing step by step instructions, I will try to answer any questions you have. 

Below I have attached the .mdat of the shifted IRs. That will help you see the window settings used in the final chart and also allow you to view the impact on the other charts not shown above. 

Note that the 1.137 ms TW delay in the Xilica needs to be set relative to whatever MR delay setting that may be needed to time align the W-MR XO. So if the MR has a delay of 1.000 ms to align it with the W then the TW delay should be reset to 2.137 ms in the Xilica. Of course it may well be the W that needs the delay added. In that case the MR delay stays set at 0.000 ms and the TW to 1.137 ms.

View attachment ja2 Philippe75 MR-TW 1.mdat


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## Philippe75

Great job! You mentionned years of experience, I will have to be patient 

Starting from the same original file, I managed to start identifying your individual steps. But I will need some time as I am not that trained with REW. Here are some "basic" questions:
- Once impulses are shifted to 0 ms, the Window Ref Time stays at the original Ref time 5,8 ms. Therefore, I can not adjust the IR Windows properly to "generate" a suitable phase curve. How did you manage to remove the Window Ref Time?
- You are using different methods in IR Windows, left is Turkey 0.25, right is Blackman-Harris 4. What are the major differences? Window slopes? If you have links that present the different methods, I would be very interesting.

Once I will be able to remove delays properly within REW, I will be able to work on phase tracking and aligning, understand the basics of the simulation and try to find by myself your very aligned curves!!! Again, I am really impressed.:clap:


In regards to mic position, height is not far from ears height at listening position. The sofa is low ;D...


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## jtalden

Philippe75 said:


> Starting from the same original file, I managed to start identifying your individual steps. But I will need some time as I am not that trained with REW. Here are some "basic" questions:
> - Once impulses are shifted to 0 ms, the Window Ref Time stays at the original Ref time 5,8 ms. Therefore, I can not adjust the IR Windows properly to "generate" a suitable phase curve. How did you manage to remove the Window Ref Time?


Click the "IR Windows Box" on the REW main panel. The Pop-up allows changes to "Window Ref Time" as well as all the other settings. We must select "Apply Windows" to activate any changes. I am always forgetting this step. :doh:



> - You are using different methods in IR Windows, left is Turkey 0.25, right is Blackman-Harris 4. What are the major differences? Window slopes? If you have links that present the different methods, I would be very interesting.


I don't have links, but general info is easily found. I just Google it. Wiki usually seems to cover these types of things pretty well.

The left window function is not critical. I like to leave it at the default Turkey 0.25 for phase work. Just place the window time just a little before the IR rise. The right window function is helpful to weight the IR data. The weighting starts at 100% and tapers off to zero with the shape dependent upon the function selected. REW allows us to click the "Window" box at the bottom the "Impulse" panel to show the shape and timing of the window settings selected. We need to be careful with the setting of the right window. We do not want to distort any data at the frequencies of interest. The SPL will be distorted before the phase is adversely affected. The minimum right window time I would reluctantly use is:

Right Window Time = 1000 / Fm * 2 (ms) 

Where "Fm" Is the minimum freq of interest for the analysis. So if the min freq we are interested in is 500Hz then the right window setting is no less than 4.000 ms. Using a window function that tapers off quickly with this setting may cause too much distortion for some purposes.

In general the right window should be kept as large as possible. Reduce it from the default 500 ms only the amount necessary to clean up the chart while being careful not to distort the range of interest. It takes some experimentation and practice.



> Once I will be able to remove delays properly within REW, I will be able to work on phase tracking and aligning, understand the basics of the simulation and try to find by myself your very aligned curves!!!


Tips:
Load the MR, TW voices into REW twice (2 copies). We can also use "(A + B) / 2" to copy the 2 measurements.
Do not shift the first pair. Leave them at the starting positions (reference positions).
Move only the duplicate pair. That will allow a measurement of the change in IR positions in case we lose track of the adjustments we make. 

*Important: Do not use any smoothing filter beyond 1/48 octave as the phase chart in REW gets very hard to read.*

1) Move the TW IR peak near 0 ms. (do not move it again)
2) Move the MR the same exact amount. (the Phase overlay chart then shows the initial phase tracking)
3) Adjust the MR IR again (and again) as needed to improve the phase tracking. 

After finding the best condition then Invert the IR for the MR and try again. We may find the phase tracking optimized better with the MR inverted. It all depends on the XO filters chosen and the driver characteristics. If neither of these 2 options track as well as we want, we can change the XO filter for one of the voices and try again. It takes some experience to predict the type of XO filter change that might help. There is no need to keep the LPF and HPF the same type. We can mix the filter types used in the XO as needed to meet the overall requirements, i.e., speaker protection, phase tracking, directivity control, etc.


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## Philippe75

:bigsmile: We always forgot to click on apply!

Seriously, with given measures I am able to identify delays in between speakers to setup Xilica parameters accordingly. The Tweeter is indeed to be delayed as it is physically in front of the Beyma (confirmed by measure). Average delay of all measures is -1,144 ms but there are little différences from one measure (one filter type and one slope) to another. I suppose this is normal, due to the filter transfer function.

But, at that point, I need to set impulses at 0 ms to get ridd of the delays for next steps. If I am right, there are numerous ways to achieve it :
- Use the Control button and then the t=0 ms offset function
- Use the Control button and then Estimate IR delay function

In my previous post, I used the Control button and then T=0 ms offset function (added offset was 5,832 ms). The impulse is now aligned with 0 but Window Ref Time is -6 ms in IR Windows (rounded I suppose). And when I use the Estimate IR delay function, the impluse is well aligned with 0 and Window Ref Time is 0 ms.

Which way is the good one to go one step beyond ?


In addition, should I generate a Minimum phase and work with it or use the measured phase after having adjusted the IR Windows (left & right) and offset ?

I suppose the best would be to work in a different way :
- Measure speakers
- Identify delays and setup Xilica accordingly
- New measure of speakers with delays => impulse are all aligned at 0
- Go on


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## Philippe75

First testing with my understanding of the step by step method (same file, Link 24dB at 8000Hz) - GREEN=Fostex, BLUE=Beyma, RED=A+B:








The step by step method built so far:


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## Philippe75

Deleted, I made a mistake... Elements are of no use.


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## jtalden

Philippe75 said:


> Seriously, with given measures I am able to identify delays in between speakers to setup Xilica parameters accordingly. The Tweeter is indeed to be delayed as it is physically in front of the Beyma (confirmed by measure). Average delay of all measures is -1,144 ms but there are little différences from one measure (one filter type and one slope) to another. I suppose this is normal, due to the filter transfer function.


Yes, I mentioned above that all delays are dependent upon the XO settings selected. Any change to the filters can impact the timing. 



> But, at that point, I need to set impulses at 0 ms to get ridd of the delays for next steps. If I am right, there are numerous ways to achieve it :
> - Use the Control button and then the t=0 ms offset function
> - Use the Control button and then Estimate IR delay function
> 
> In my previous post, I used the Control button and then T=0 ms offset function (added offset was 5,832 ms). The impulse is now aligned with 0 but Window Ref Time is -6 ms in IR Windows (rounded I suppose). And when I use the Estimate IR delay function, the impluse is well aligned with 0 and Window Ref Time is 0 ms.
> 
> Which way is the good one to go one step beyond ?


The shift to "Window Ref Time" is a new action for the recent REW Beta releases. It used to stay at 0 ms when IR shifts were made. For our purposes we ideally want it set to 0 ms. If we are using narrow window settings this is critical as we do not want to clip any data accidentally with the window settings. At the default wide window setting it would not normally be a problem even if we don't change it back to zero. Only a very large shift (>100 ms) would cause a problem. 

I had not noticed that the Window Ref Time value may not be the same depending on how an IR shift was made. I would think it would always be the value of the shift. Maybe the selection of "Preferences/Sub-sample Timing Adjustment" makes the difference? If it is not activated REW works in full sample increments for shifts. I have this feature selected (sub samples shifts allowed) and have not noticed any problems. I rarely use the automatic shift feature however so maybe that is why I haven't seen any discrepancy?

I offset the TW manually to near 0 ms. Any offset setting works from a functional perspective, but a relatively straight smooth phase trace makes it much easier to find the best overlap setting. I most often select the amount such that the TW phase chart smoothly falls to someplace near 0° as I did above. Sometimes I set it so the phase falloff is linearly falling. I think it's easier to judge a phase tracking error when using a relatively linear target line. There is no functional difference if we work with slightly different offsets and thus have more or less curvature in the TW phase target. It is only the closeness of the MR phase tracking that is the objective. I have done the job successfully with TW phase ending anyplace in between +90° to -180°. 



> In addition, should I generate a Minimum phase and work with it or use the measured phase after having adjusted the IR Windows (left & right) and offset ?


Ideally, we are adjusting the direct sound phase tracking error to be minimized. With experience and good windowing this is very manageable to do accurately looking a the phase chart. The minimum phase is calculated based on the SPL measurements. The minimum phase does help to pull out the direct phase trends from room effects. It also tilts the overall phase somewhat as a result of any level changes as those due to house curves. To the extent that the minimum phase does not track the true direct signal phase it is not the ideal choice for the job.

Since the REW phase overlay chart does not offer the option to show minimum phase, I have not looked to see if any resulting setting difference would be significant or not. My expectation is that any difference would be very trivial for this job. 



> I suppose the best would be to work in a different way :
> - Measure speakers
> - Identify delays and setup Xilica accordingly
> - New measure of speakers with delays => impulse are all aligned at 0
> - Go on


Yes, I always measure again after applying the delay setting to confirm it was properly entered. When entered properly I have never found a discrepancy between the calculated prediction and the measured result.


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## jtalden

Philippe75 said:


> The step by step method built so far:


So, you have some program management skills! 

This is very organized and a good first attempt. There is lots of details that will be different for each XO project but this is getting close for this job. The major steps are there but there are several changes/clarifications that should be made. If you want me to edit it, attach or send me a .xls or csv text file so I can do it directly. 

I could just comment on this here if you like, but it is more difficult that way.


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## Philippe75

Here it the Excel file, generic so far and it need to be adapted for each project indeed.
View attachment Method.zip


I am not too sure I know criteria to properly define the IR Window. Currently, I am looking at the shape of the phase curve at very low Right IR value (10 ms for instance) and then increasing it slowly until a first wrap appears. Any tips ?


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## jtalden

Philippe75 said:


> Interesting for choosing ideal filter types and slopes in regards of phase. The most linear ones are Link 18 dB, Butt 30 dB and Butt 18 dB (5,5 kHz to 15 kHz)


?? - You may be misunderstanding something based on this comment.

None of the traces are linear. They all have phase rotation. The phase rotation of a voice+XO is dependent upon the phase rotation of the voice itself plus the impact of the phase rotation of the XO filters involved. The phase rotation of the filters is basically dependent upon the order of the filter. There only a very minor phase difference due to the common IIR functions used; Butterworth, Bessel, or Linkwitz-Riley.

On this basis, in general, we can say (and would see in your chart) that there are only 6 groups of phase rotations; Raw, 12, 18, 24, 30, 36 dB/Octave.

I said "would see" because there are 2 different offsets that occurred when your chart was unwrapped. This broke the chart into 2 sets of data spaced 360° apart. This is one of the confusing risks when unwrapping phase in REW. It unwraps it at the cursor position as you noted above, but it is easy to get mislead as several things can create problems. The largest problem is the impact of room reflections and standing waves that create an apparent 360° phase rotation over a very narrow freq range. To unwrap a series like this correctly the cursor needs to placed at a freq near the XO and where all the traces are together. In practice it is only practical to unwrap phase when all the room reflections are eliminated in the freq range of interest by using appropriate window settings. It looks like you used a right window that was too large. I think I was down to about 8 ms to avoid any reflections on the chart. It was tighter than I normally use but I wanted a perfectly clean chart across the entire displayed range to avoid confusion.

It may be is safest for the beginner to leave the chart as "wrapped" and just learn to ignore the rapid jumps in the phase response of 360°. I usually stop at the first phase chart I provided above. It looks a little messy, but it is not really that hard to read. I only offered the second chart to clean up the lines and center them so that it is perfectly clear how the phase tracks for those who have not yet learned how to properly interpret the original chart.


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## jtalden

Philippe75 said:


> Here it the Excel file, generic so far and it need to be adapted for each project indeed.
> View attachment 53514
> 
> 
> I am not too sure I know criteria to properly define the IR Window. Currently, I am looking at the shape of the phase curve at very low Right IR value (10 ms for instance) and then increasing it slowly until a first wrap appears. Any tips ?


Yes, I do that sometimes also. Reduce it as needed to avoid any wraps due to reflections and then increase until the point just before the first wrap occurs. Make sure that the setting is large enough to avoid distortion in the traces. Sometimes (often when measuring a W-MR XO from the LP) that is not acceptable. We then just need to leave a larger window and ignore the spots where a reflection occurs. In these cases it is also not possible to use "Unwrap" as that will jump an 360° additional at each additional reflection. That destroys any chance of reading the phase tracking properly. It's best to learn to read the chart with "wrap" applied. Would it be helpful If I marked up a reflection plagued phase chart to show how to pull out the direct sound phase path? I could use my first chart above or you could provide one that you think would help.

Below are my changes to the step-by-step. I am not sure they will really help. We all have a different way of understanding these things. When learning, I think its best to keep this process separate for each XO. I therefore dropped reference to your woofer. Instead I used "HF driver" and LF Driver" There are always only 2 drivers involved and the higher freq one is always the reference one for the process. The same steps then apply for your W-MR XO or to anyone else using different driver designations.

I left out the steps regarding window settings to the phase chart. It is too specific a situation to be very helpful. The general guidelines already provided are about as clear as I can offer. We can think of this as a subroutine to be uses as needed to better define the direct sound phase trace.

I also didn't add in steps regarding the SPL A + B and identification of the XO range. I assumed those steps were clear to you.


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## Philippe75

> So, you have some program management skills!


Yes, senior project director in IT field. I like things to be clear, understandable and applicable. Even for my hobby.....:rofl:


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## Philippe75

OK, I am fine tuning the step by step method. When I will be done, I post it again if this can help and if you don't mind as this is your method.

In regards of Windows IR (gating), I was using this site to calculate floor/ceiling reflection and thus first reflection and gating in ms: http://mehlau.net/audio/floorbounce/

Since I measured speakers at 1m distance and 88cm height, first reflection is arriving at 2,98ms. So I am gating at this very very short timing for phase working. Isn't that too short ? When looking at impulse response, it seems to make sense (first echoed pop)... Too short to work on SPL however.

In regards of phase curves, and as I mentionned earlier, my graph was not right because of gating errors. Do you see any weird things below (Fostex/Tweeter=8000 Hz, all filter types, gating at 2,6ms as an average)? Curves are starting at different angles depending on filter type and order, but ther are aggregating in the end...









In addition, and if I am right, starting from impulse at 0 ms : when I am changing impulse by +X ms, phase curve is going "up" (+xx° - delay is reducing) whereas when I am changin impulse by -X ms, phase curve is going "down" (-xx° - delay is growing). Am I right ?


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## jtalden

Philippe75 said:


> OK, I am fine tuning the step by step method. When I will be done, I post it again if this can help and if you don't mind as this is your method.


Okay.



> In regards of Windows IR (gating), I was using this site to calculate floor/ceiling reflection and thus first reflection and gating in ms :
> 
> Since I measure speakers at 1m distance and 88cm height, first reflection is arriving at 2,98ms. So I am gating at this very very short timing for phase working. Isn't that too short ? When looking at Impulse, it seems to make sense (first echoed pop)... To short to work on SPL however.


This is generally too short for phase tracking in my opinion. It may be okay for a HF XO. We can tell by increasing the right window and see if the phase shifts position in the range of interest.

This mic position and measurement technique is normally intended to window out the first major reflection, the floor bounce, in order to see the SPL that results for the frequencies above that. Many designers are looking to present quasi anechoic SPL response for the speaker without the room influence, but with baffle step and other very early reflections. They most often just measure on the TW axis because they do not know what the geometry of the room setup will be. This is not the purpose of our measurements however.

Even for that purpose it would probably be better with a large speaker like yours to measure at a 1.5m or even 2m so the sound from the 2 voices are more fully merged. 1m is far enough for away for smaller speakers that have the TW to MR center-to-center pretty close and the XO freq nearer 2kHz. 

That whole quasi anechoic SPL thing is another subject however, so let's get back to our purpose; phase tracking.

The phase tracking can also be done at a 1m distance, so long as the mic height is on or very near the listening axis for the 2 voices being measured. 

If we use the linked graphic for our purpose, we need to add a little more detail.
Note that the graphic does not show 2 voices and for our purposes h1 would then be between the 2 voices vertically. So, if the height to the TW center is 90cm and the MR horn center is 60cm then h1=75cm. 
If our LP is h3=88 cm at d2=3m distant and we are measuring at a third that distance (d=1m) then the mic height would ideally be h2=75+[(88-75)/3]=79.33cm to place it on the listening axis vertically. It obviously doesn't need to be that exact, but I would keep the angular error pretty small vertically. I think I suggested ±3° above for your speaker type and XO. Possibly you followed this logic previously, but you referenced link did not seem relevant so hence the recap. The easy way is just place the mic at the LP.

For phase tracking we are interested in the direct sound phase response, but we don't mind mentally interpolating between the phase irregularities that are caused by the room reflections. We will just discount the areas of reflections and mentally connect the areas where the signal is clear. This can often be done without any windowing with some practice. Windowing does help to reduce the number of the reflections displayed so it is helpful particularly when starting out. We Just reduce the right window more and more until we can see the tracking of direct sound phase. 



> In regards of phase curves, and as I mentionned earlier, my graphs were not right because of gating errors. That is why I thought some XO filters & slopes were more "linear" than others. But as you mentionned, only the order of a filter is a correct "group by" aggregator of phase curves... I continue to work on this to understand...


Just remember that unwrapping the phase can easily go wrong so I would suggest that you first stick with wrapped phase and learn how to read that.


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## jtalden

Philippe75 said:


> In regards of phase curves, and as I mentionned earlier, my graph was not right because of gating errors. Do you see any weird things below (Fostex/Tweeter=8000 Hz, all filter types, gating at 2,6ms as an average)? Curves are starting at different angles depending on filter type and order, but ther are aggregating in the end...


Additional comments follow based on the edited version.

Regarding the chart:
I mislead you before. I assumed the 2 groups were offset 360° due to unwrapping and I can now make out the scale as see they are separated by only about 180°. I also was misleading in my logic as what is to be expected. Sorry about that.

I am no longer 100% confident in how best to present and interpret this set of data. I would have to see the file so I could shift the IRs where I think they should be located to be sure how best to characterize the situation. I suspect I would shift each IR such that the passband of the TW (maybe 12k-17k) is as level and flat as possible and then look a the differences. Each higher order should shift the passband down 90°. Also the stopband (maybe <10k) will shift steeper up. :justdontknow:

This does illustrate the general idea mentioned above that changing order of the HPF will impact the how well the phase will track the lower voice. I guess my lack of clear understanding is the reason that I often change the wrong filter or the wrong direction and have try again.

If this is important to you I can try something similar on my setup and clarify the logic in my mind. I see this as a learning experiment (I do lots of those and have done this one before, but its been awhile). It does not directly relate to the job at hand however unless you want to try to reduce the small tracking error at the lower end of XO range and want to be more sure which HPF setting might be most beneficial. A little trial and error is easier for me.



> In addition, and if I am right, starting from impulse at 0 ms : when I am changing impulse by +X ms, phase curve is going "up" (+xx° - delay is reducing) whereas when I am changin impulse by -X ms, phase curve is going "down" (-xx° - delay is growing). Am I right ?


Yes, you are correct. +xx ms shift for the lower voice corresponds to reducing the relative delay of that voice in the Xilica. The lower voice phase curve is moving up at the top end of its range relative to the upper reference voice. This is a little confusing at first. Actually, I still often get confused on direction when making adjustments for LP distances or filter order changes.


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## Philippe75

As for you, the analysis of HPF effect on the tweeter was to visualize resulting curves with existing measures (and thus limits : mic at 1m distance, 88 cm height (middle in between woofer height and tweeter height). I have done the same with MW again but I did not post it on the forum. I need to enlarge gating and see if indeed I can work with slightly wrapped curves. Learning curve is tough...

It could be interesting to compare with your measures indeed (if this is not a problem for you) and then share findings for next steps.

I plan to do new measures this next WE following your instructions: mic at 2 m distance, LP height and aligned with LP to speaker direction. But I will focus on a first project: MF-TW XO at 8000 Hz (seems to be a good starting point) and WF-MW XO at 650 Hz (higher than initially expected). I will try some combinations with 12dB, 18dB and 24dB slopes because higher slopes have more impact on phase even on the listening range.


I maybe need to clarify one point if needed: I am focusing on fine tuning the speaker itself more that working on room effect at the moment. I thought it was good to get the best of the speaker first (and thus quasi anechoic analysis) and then see how to improve the result because of the room. But I should maybe consider the whole thing as 1 unique and global objective. And I think this is what your are telling me. Am I right ?


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## jtalden

Philippe75 said:


> It could be interesting to compare with your measures indeed (if this is not a problem for you) and then share findings for next steps.


I will do that. It may take a couple of days. I think there are several ways to depict the relationship. The way I prefer may not be the way you prefer or the way the textbook depicts it. *Here* is a good link to some insight on XOs and the impact on phase. 



> I will try some combinations with 12dB, 18dB and 24dB slopes because higher slopes have more impact on phase even on the listening range.


Just FYI: Yes, the total phase rotation and GD is increased with filter order. The AES studies that I have seen experts refer to apparently show that phase rotation and GD within the limits that we are using with the Xilica does not have an impact on sound quality. Even with selected test signals to highlight a problem the impact was not significant. [I think my understanding is on this point is reasonably correct.] I have tested this myself using ABX testing using earphones to listen to some of my music with and without over 720° phase rotation added. I could not differentiate the 2 files. My hearing is substandard though (nasty tinnitus) so I can only say that I personally was not able to detect it. There are many other who claim to hear it, but I suspect that have not really tested it a reasonable way?

Since you are using JRiver MC you can use it to automatically take out all the phase rotation in the final response if you like. I do that even though I can't hear the difference. I just like the look of a flat phase and GD response and the cleaner looking square wave response. That would be the last step in the process however and we would not need to worry about that now. It does however negate any worry over using higher filter orders.



> I maybe need to clarify one point if needed: I am focusing on fine tuning the speaker itself more that working on room effect at the moment. I thought it was good to get the best of the speaker first (and thus quasi anechoic analysis) and then see how to improve the result because of the room. But I should maybe consider the whole thing as 1 unique and global objective. And I think this is what your are telling me. Am I right ?


There are many who optimize the near field response down to the freq level possible - maybe 500 Hz? That is the only option if the application is not known, so those that sell speakers tend to insist that this is the best way. I have tended to look at the response in my listening area (average of a window around the MLP and EQ to that. I have since looked at the near field results of that approach. At least in my room and setup the SPL response (and required EQ) >500Hz is pretty much the same either way. It's pretty clear (little disagreement) that it is best to EQ the bass range (<200Hz ?) to the LP measurements. The more open issue for me is the 200-500 transition range. I have no clear EQ preference for this range and see lots of irregularity in my setup. Sometime I leave it as is and sometimes I smooth it out to some extent.


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## jtalden

I tested the effect of the LPF filter settings on my TW (SEAS DXT). The Mic was about 1m and I left the current 2 EQ filters that shape the response of the TW activated. I moved the normal XO point up to 3.46 kHz to allow plenty of safety for the first order filter measurement. I considered using loopback measurements directly from my DCX, but that is a little more setup effort for me so I decided against it. I will leave that job to you if you want to see the impact of the HPF without the impact of the mic and TW. 

In this first series of charts I aligned the IRs so they all target 20kHz at 0°. The delay to the initial IR rise is thus about all the same time as the highest frequencies arrive first for IIR filters.

SPL (Same for both sets of charts)








IR alignment:








Phase result:








We can see the phase slope is different for each order and the impact of the filter function is small. This is probably the best way to present the data for basic understanding.

Below I aligned the IRs to provide the flattest phase in the passband (7k-12k Hz). This way we see some shift to the phase of the passband although it is not and much as I remembered and the phase shifted up with higher orders, not down as I stated above. There is 7 order range represented here (1st to 8th order). The total bandpass offset is about 190° so the bandpass offset for each additional order is ~27°. That is not an expected number for me and not what I remembered from previous investigations so this may be dependent upon the other conditions of the test setup. This is why I think the first way of aligning the IRs may be a better way to draw a generalized understanding. 

IR alignment:








Phase result:








I just remembered that I forgot to disable a 20kHz But-24 LPF filter that I use to filter out any ultrasonic breakup resonances from my aluminum dome TW. I should have noticed that as the phase is sloping up significantly above 12kHz. Some of that rise may be due to the voice coil inductance?, but most is due to the filter. I am not sure if there is any impact to the results here. I don't think it would make a significant difference, but I am not going to retest to find out - another job for you I guess. :devil:

I hope this helps.


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## Philippe75

I could not understand this 180 deg variation in my work, especially once your testing results were published... So I took again a deep look at the file (tweeter all XO), and focus on IR gating and 0 ms alignment:
- some impulses were reversed and/or not well aligned => corrected
- gating was too short => -1 ms left (Turkey 0.25) and +31 ms right (Blackmann).

Initial alignment was done with function "Estimate IR Delay" and then "Shift IR": impulse "top" is never at 0 ms. *0 ms alignment is to be done very very carefully and impulse is to be well oriented before working on curves.* We could easily lost a lot of time working on inappropriate data if we don't control this critical element!

Tweeter - Result is much better, all phase curves now have the "same" deg value at 20 kHz:















Mid-woofer is also much better:


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## Philippe75

Just to inform you that using JMLC Excel sheet simulator, we are able to generate HPF & LPF combinations quite easily (Link, Bess & Butt - 6 dB to 24 dB). From this basis, it is simple to extract data to generate House curve files (as many as needed). Once loaded in REW (one for each simulation), we can indeed use PEQ simulation to adjust SPL curve.

However, it is still not as friendy as an internal HPF and LPF XO generator. :bigsmile:


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## jtalden

Philippe75 said:


> Initial alignment was done with function "Estimate IR Delay" and then "Shift IR": impulse "top" is never at 0 ms. *0 ms alignment is to be done very very carefully and impulse is to be well oriented before working on curves.* We could easily lost a lot of time working on inappropriate data if we don't control this critical element!


Agreed. Phase charts are very sensitive to proper IR alignment. The higher the freq of interest the more sensitive it is. We are normally looking for relative delay between voices rather than this type of HPF order comparison. For Phase Tracking charts we are not changing the filters. We use REW loopback timing to provide the accurate time alignment. We didn't do that for this purpose because the different filter orders will have different delays and thus the IRs would not be aligned as we want anyway. We knew we would need to manually to align them to best compare the phase slopes of the different orders. 



> Tweeter - Result is much better, all phase curves now have the "same" deg value at 20 kHz:


There is still a problem with this phase chart. You have 5 orders measured and only 3 groups of phase traces. There should be 5 groups. All the 2nd order traces need to be overlaid with each other. The same for the 3rd, 4th, 5th, and 6th orders. The orders should not overlap each other. Your chart should look like mine did. I marked the groups with the order of the filters that created them. 

It looks like the problem is still that the phase is not well enough aligned at 20kHz. There appears to be almost a 90° difference in some of them. [It may be easier if the phase line is more horizontal where we are doing the alignment. The steep slope at 20kHz is probably not helping the job of alignment. I would have shifted all the IRs +0.025ms or even maybe +0.050ms such that the phase crosses 20kHz more horizontally.] 

Shifted or not, the fine tuning of the phase adjustment needs to have all traces cross 20kHz at the same point. I did that on my first chart to the nearest 0.001ms. I suggest you try again.

If you continue have problems, and you want me to organize it for you, I can do that. The file is probably too large to post here; even zipped. You could break it into smaller groups, email me a .zip, use a drop box, or ?? 



> Mid-woofer is also much better:


This set looks okay for the 650Hz HPF settings. The phase traces are pretty well aligned at 8kHz. At least well enough for good accuracy of the low range. We see 5 groups of phase traces at the LF side. If these represent the 5 orders of filters then it looks as expected. 

Changing the order of a LPF has little influence to the job of phase tracking because the slope of the stopband phase can be adjusted as needed by just changing the delay.


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## Philippe75

Well, I worked again on XO effects on phase and there were remaining errors in IR alignment. I corrected these and aligned phase curves at 20 kHz. Indeed, phases are group by filter order and filters have limited effects on phase at high frequencies but more at low frequencies (especially when working on WF measures). I will stop there.


I did some new measurments this WE. It was difficult because the weather was beautifull and I had to work in the garden  The project is based on the filter mentionned in the post "Aligning driver phase": *LPF=Bess-24 dB, HPF=Butt-12 dB with XO defined at 650 Hz and 8000 Hz.* Mic is at 2 m distance and 96 cm height (ears at listening position), still pointed towards speakers. SPLs have been corrected (dB offsets), results are as following (*Red*=WF, *Blue*=MR, *Green*=TW, *Pink*=MR+TW). Keep in mind that WF-MR is still "work in progress", do not pay too much attention to it:















I started by working on* TW-MR*. IRs have been aligned and adjusted as well as polarity. Simulations given below have been confirmed live, they are equal. Good point!















As mentionned, I am still working on *MR-WF*. It is far harder to proceed as WF phase is really weird. I setup a very large IR window (right is 100ms) and unwrapped phase (will have to work on wrapped phase as well, ,ext step). Not too bad but I need some advice.








In coming days, I will improve this first result by testing and fine tuning parameters. I also want compare this project with different measures done the same day and see if there is a better result when slightly moving the WF-MR XO (600 or 700 Hz with 8 000 Hz) or MR-TW XO (7000 or 7500 with 650 Hz). I also need to improve how to use unwrapped phase which, so far, remains difficult to read and so make the good decision.


Here is the updated "Step by step" file filled with values of the project. There are now 4 sheets:
-Raw measuring: how to measure a speaker without any XO nor delay or whatever
-XO measuring: how to measure a speaker with HPF and/or LPF
-XO optimization: what we are discussing
-SPL optimization: room has been provided for the next step if required
...and an additional column called "Value" that is to be used to store settings or results. This column is also used to calculate results such as speakers global delays. The method remains oriented to my setup, but it can be easily adapted. I still need to finalize WF-MR section and add rooms for IR offset vs Inverted phase optimization method (they are still mixed). Finally, I will work on the SPL optimization sheet to store PEQ generated by REW EQ to be applied on speakers/outputs or inputs.
View attachment XO - LPF=Butt-12dB-650Hz HPF=Bess-24dB-650Hz LPF=Butt-12dB-8000Hz HPF=Bess-24dB-8000Hz.zip


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## jtalden

I got bogged down in the detail of your spreadsheet. I don't think I can help much as I have a different way to think of this. Your version may work for you as there are many ways to travel in the process that results in successful phase tracking. 

I think of REW as a heavily loaded tool box that is very flexible in the jobs it can analyze. Specify the job and then select from the REW tools and settings to work through to an answer. It is best to understand the tools/controls well enough to resolve problems that occur. Phase tracking is tricky analysis and the REW features and settings may very significantly for different situations. A detailed step by step is thus pretty specific to a single job. I will attach a spreadsheet that comes a little closer to my thought process in case it helps.

In my spreadsheet I tried to capture just the major steps needed to do the job generally. Each step may require familiarity with different settings and features. The steps are not detailed as they are generic, not specific to a particular job. I don't know is this will be helpful then for your thought process.

Comments on the content of your spreadsheet:
> I assume the purpose of Tab 1 process is just to identify the bandpass range of the voices so XO settings can be selected? An option is to just use the manufacturers published SPL response curves to decide on the initial XO settings. 

> Regarding measuring you often note; "Execute Check levels and adjust Accuphase volume for 12 dB Headroom".

No - The measuring volume can be set for 75dB. Just adjust the Accuphase volume so the average of bandpass portion of the trace is someplace near 75dB. [The headroom is not a factor when making measurements. The resulting headroom is established by the REW SPL meter calibration process done at REW setup. It does not need to be considered for acoustic measurements unless it is found to be insufficient, i.e., clipping occurs, or the level of the measurement is reported to be very low (maybe <-35). [In either of these cases the REW SPL meter calibration process should be repeated after the appropriate adjustment to the input level on the Tascam is made.] There is no need to check levels before each measurement.

> It looks like you are creating a very large .mdat with all voices involved. That gets tough to manage. I would break it out into small (2 driver measurement) working experiments. One experiment for each of the 2 XOs. See my spreadsheet.

> I am not sure why you are looking at all the different filter possibilities. That's lots of work. Just select the XO you would like and determine the delay. Then adjust the HPF filter order only if it is not possible to get good enough tracking with the initial choice. I mentioned that with your wide overlap between drivers an LR-24 is a good and popular candidate. Some insist lower orders are better so you can choose what you like. Some selections will track better than others. Your initial LR-24 selection worked out very well for the 8k XO. It won't get much better than that.

> The choice of the filters in my "how to" thread was just a current example at the time intended to show that the filters do not need to be the same in the XO for excellent results. I probably only used that particular setup for a few days. My SWs-MW XO filter selections are always completely different from the MW-TW XO filters. I am not sure why you retested your voices using those settings for both XOs. Each XO will be optimized with different settings as the voices involved have different design characteristics.

> I didn't review Tab 4 regarding EQ adjustment as that is a completely different subject.

> If you want me to offer a recommendation for your W-MR XO delay just provide a measurement of the 2 voices in the left speaker and also in the right speaker using your proposed XO settings (4 total measurements). I will find a delay that provides the best tracking. You can see if that agrees with your findings. I would need to see both speakers as the location of the speakers can influence the phase traces significantly for lower freq XOs and I would be looking for a single XO setting that best serves both right and left speakers.

View attachment ja-PhaseTracking Process.zip


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## Philippe75

The excel file is slowly becoming a TO DO list for one specific objective as there are a lot of parameters to take into account. It is also a way to keep a trace of parameters at the time of the action, even if some parameters may become fixed over the time... This is why it is so detailed.

I fully agree on your vision of REW: a very powerful tool box that can be used in several ways! And as you mentioned, there is not only 1 way to obtain the good result. However, I noticed that some actions have to be done in the correct order. Therefore, the file is a translation of my understanding of sequences with my logic which is very probably different from one to another. But, if the result is correct and if I did not forgot steps, then it is good! If this can help others to start (as you did with me) and adjust it to their own logic/approach, then it is perfect!


For your information, *RED* cells are to be input, *BLUE *cells are the calculated ones.

The first tab is intended to speaker measurement in raw listening conditions. This is why each speaker has its own parameter set (e.g. mic distance, mic height…). As we can see, there are always slight differences between manufacturer curves and in-house curves (remember the 13 kHz peak in MR+horn combination). It is also a convenient way using REW, to compare the original curve with one XO curve. I personally often take a look at the original curve of a speaker to visualize effects of the XO and ... learn. This step is of course not mandatory, and should probably be done once for all.

The second tab is intended to XO measurement in listening conditions for all speakers. This is why some parameters are defined for all speakers (e.g. mic distance and mic height) and others by speakers (e.g. HPF XO type, slope and frequency). We mays not need to change some parameters project after project, but the file manages it if needed. Of course, all XO combinations do not have to be measured but just the one we want to implement. The LPF=Bessel-24dB and HPF=Butt-12dB XO was a good example to test the file. One can indeed choose to implement different WF-MR and MR-TW XO, I will probably do so once I had defined the good XO for WF-MR. Next project…

The third tab is intended to XO phase optimization using delays and inverted IR if needed. It is still under Work in progress. Based on initial speakers IR delay and additional speakers delays generated by the optimization process, it computes each final speakers delays to be implemented in the digital crossover (i.e. the one in between speakers minus the remaining global delay). There are indeed a lot of measures within one “.MDAT” file. However, it can split it for WF-MR and MR-TW if needed

The last tab is intended to SPL optimization once XO are defined and phase optimized. Do not pay attention to it right now, as we may need to discuss things in detail later depending on the objective (room equalization, speaker equalization...).


In parallel, I try to improve my usage of IR window and Wrapp/Unwrapp function as they appear to be very sensitive. And as you may know, garbage in-garbage out…


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## jtalden

:sn:


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## Philippe75

Unfortenately, I did not had the time to do all what I wanted during WE. I am still working on the TO DO file (RAW measurement is now in a separate file as it is a specific need and I integrated minor changes) and testing it. Thus, I found some strange results while re-applying REW steps for a second time on the same project (starting from scratch except the same .MDAT files). Probably due to my limited understanding of "how to use wrap phase" and IR delay measurement... 2 simple questions.

Do you have any tips to properly measure IR delay? Currently, I am opening the Impulse tab and measuring the delay at the top of the IR curve. But little shifts (+/- 0,001 ms) have limited effects on IR that are hardly measurable, especially on short impulses such as TW.

How can I be sure that 2 wrap phase curves cross at the same angle at a specific frequency? I managed to have WF and MR phase curves crossing at 650 Hz using IR shifting, but how can I be sure they are really crossing without +/- 360° effect? In that exercise, Unwrap curve is much more convinient...

Thanks for your help.


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## jtalden

It is very difficult to provide a description that goes beyond what has been said here and in the other *recent thread* on this subject. The best approach is to post an .mdat file with the LW, LMR, RW, RMR and I can try to show how it applies to your situation.

That said, here is another attempt to provide helpful XO timing guidance when using an IIR (minimum phase) XO like the Xilica:
> To see the phase of the direct sound through the noise of the chart, we need to know the expected shape of the phase trace for the 2 voices. The expected shape is the same as would be measured if the mic was placed very close to the voice to eliminate the influence of the room. 
> The initial rise of the IR is a good indicator of when sound first arrives as the mic. 
> The first IR peak is the arrival time for the highest measured freq for that voice. The lower freqs in that measurement arrive later (delayed) and cannot be seen in the IR. 
> If we aligned the initial IR peaks of your W and MR we would be effectively be matching the timing of the W max freq (maybe 1kHz) with the time of the MR max freq of the MR (maybe 16kHz). This is not the correct timing.
> If we initially align the initial rise of the 2 IRs we will be close to the correct timing. This will place the peak of the W behind the peak of the MR. The arrival time of the MR's lower freqs will be closer to the arrival time of the W's Higher freqs; closer to what we need for the XO range freqs to align.
> We can then look at the overlay phase chart (1/48 octave smoothed only) and make the final minor MR adjustment to optimize the phase tracking of the 2 voices.


[Doing it this way (overlay phase chart) is the only way to actually see how good the phase tracking is and to be assured that some other setting of the XO filter, polarity, or delay will not result in a better alignment. This rigor is not required to get good results however. If we start with the initial IR timing recommended above then it is usually easier to make the final adjustment using either RTA or sweep measurements with minor MR IR adjustments until the SPL is maximized in the XO range.]


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## Philippe75

OK, I'll do that. I confirm that WF phase is very different from original driver's phase when aligned with MR on peak. This is less visible with MR and TW, waves length being much comparable. That is why I was probably confused with phase shape (especially tails) and window IR definition.

Thanks for the link to the recent post, very interesting. I need to have a much closer look to all posts...

Thanks again for all your time given to new users like me! This make me think we should build a FAQ on the forum with such shared best pratices.


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## Philippe75

OK, I restart simulation following your recommendations. I took the opportunity to improve again the STEP BY STEP file as it was not in line with what I was really doing. Seems I am a good testing user ??? 

However, just a simple question. In regards to our last conversation about 0ms IR alignment, the following IR seems fine (MR):








What for the following slope (TW): should I consider 0ms at 1) 0%, 2) at the top of the small initial slope or 3) 0% but after the small initial slope?








So far, I considered 0ms at 1) 0%...


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## jtalden

"1" fits my description of the starting place for phase tracking adjustments. In my somewhat limited experience with various setups that starting alignment seems to work out reasonably well. 

Because of the variability in XO settings and driver characteristics we can often find that initial rise to be hard to locate. The MR IR hints at the issue. There is some minor early ripple that is best ignored, as you did. I would have picked the same starting alignment spot that you did. Sometimes it get more difficult to decide. Since it is a starting point we should eventually find the right positioning no matter if the initial guess is a little further away.


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## Philippe75

Following are files of my learning project.

The TO DO excel file is filled with keyed-in values (*red cells* - based on REW findings) or calculated values (*blue cells*). This project is still based on Bess 24 dB and Butt 12 dB at 650 Hz and 7500 Hz (looks better than 8000 Hz).
View attachment XO Definition - LPF=Bess-24dB-650Hz - HPF=Butt-12dB-650Hz LPF=Bess-24dB-7500Hz - HPF=Butt-12dB-7.zip


MDAT files contain TW, MR and WF data after phase optimization. All steps are described in TO DO. No PEQ were implemented, I haven't work on this point neither on room optimization so far.
View attachment XO Definition - Optimized TW.zip

View attachment XO Definition - Optimized MR.zip

View attachment XO Definition - Optimized WF.zip


Basically, TW was the first driver I worked on, phase right tail has been "aligned" at 0 deg at XO and beyond when possible. Then I worked on MR phase to cross TW at XO frequency and finally WF to cross MR at XO frequency. I tried to work on wrapped phase by adjusting IR Window so that the phase was readable around XO without miss interpreating (hopefully).

Thanks for your feedbacks on everything that could help on improving again the result or TO DO.


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## jtalden

Philippe75 said:


> Thanks for your feedbacks on everything that could help on improving again the result or TO DO.


I reviewed the data. You now have a good working understanding of the objective and the process to best achieve it. You correctly identified and adjusted the 2 XO ranges and did not get distracted by phase error that occurs outside of the that range. This is a very good job. You are a quick study.

The delay timing that you found is very, very good. It is not possible to do better with these XO settings. Improved measured tracking results may be possible with different XO settings (see below), but would not likely result in any detectable sound improvement. It sounds like you may be intending to set different slope XO filters as LR-24 or other. You may indeed find a sound difference with this type of change as the sound field distribution in the room will change with the change in voice overlap range. Filter slope changes like this will also impact the delays needed. 

Below is the calculated overall SPL and phase of your current solution. The SPL is smoothed to 1/6 octave to better see the XO fill. I again marked the direct sound phase path to help those that are still learning. The other trace aberrations are the result of room effects and the chart "wrapping" (vertical black dotted lines). If you apply the voice delays you identified in the spreadsheet and measured the system again you should obtained these same results.

















If you are not intending to change filters, the next steps can be:
> EQ for your room/speaker setup to your chosen house curve.
> Use RePhase to design a phase correction IR filter for JRiver so that system phase is linearized (optional). In my situation, I could not identify any difference with the phase linearized. 

FYI, if you are instead interested experimenting and chasing optimal settings, some minor improvements may be possible at this step. Some options for this XO scheme are:

> Note that the MR and TW SPL levels are slightly higher than the W. It is likely that final listening testing will lead you to a house curve that slopes/curves downward 0.5-1.0 dB/octave (see the old "Bruel & Kjaer EQ curve" and *Wayne Pflughaupt's HTS technical articles* regarding "House Curves"). There are many other house curves in use and your version will need be developed to suit your music and preferences. The B&K curve is good starting place. In anticipation, you may want to adjust your voice SPL level settings a little more to better follow the expected house curve. This may ease the number and magnitude of the EQ filters that are needed.

> Note that your MR-TW XO range is supported very well just as we desired, but the SPL is somewhat high throughout that range. An option is to spread the LPF and HPF such that the boost is reduced in that range, e.g., LPF 7kHz? and HPF 8kHz?. The delay should probably be checked if a change like that is made, but it is normally a very minor impact. Another option is to change the filter functions. The different filter functions have very little impact on the phase, but do change the SPL in the transition area noticeably. This type of minor adjustment is why the LPF filter is my example was Bessel and HPF was Butterworth. I just used the Bessel filter to reduce and smooth the SPL level a little in the XO range just below the XO point. Its SPL rolls off a little sooner than Butterworth and was slightly better suited to my particular voices. 

Please understand that all these comments (excluding a change to filter slopes) would not be expected to change the sound performance assuming the same SPL EQ target is achieved. They are only helpful to minimize the number and magnitude of EQ filters necessary to meet the desired house curve.


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## Philippe75

I will not implement this XO which was good for testing purposes. As you mentionned, SPL is not really good around TW-MR XO...

When I look back in time, there were less problems with Link 24dB at 8000Hz. So I will start my first real project with Link 24dB around 7000 Hz to 7500Hz (8000Hz seems a bit too high). I also read that this was an ideal XO for tweeter... In regards of MR-WF XO, I will start with the same Link 24 dB at 700Hz, slightly higher because of the MR issue at 685Hz. I know I can use different XO types and slopes for each HPF or LPF, but currently I have no idea where to start from... I may test later different XO, but it is too early. If you have any proposal, I would appreciate.


Later on, I will concentrate on PEQ. But I must read technical articles first as I never really worked on this subject before. In addition, I understand we have different approaches and you may have the good one! I was focusing on optimizing individual driver SPL using PEQ to obtain a good overall final SPL (TW+MR+WF). I understand you are considering the final SPL as a whole that is to be optimized with all available tools first (XO type, slope, frequency for each HPF & LPF) . And then, SPL is to be optimized globally using PEQ (moderately). Technically, it means we should be using PEQ on Input (all drivers) where I was considering using PEQ on Output (individual driver). Am I right?


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## jtalden

Philippe75 said:


> I may test later different XO, but it is too early. If you have any proposal, I would appreciate.


I would agree that that the LR-24 XO is a very good compromise. It's not too shallow and not too steep. I would recommend it as the initial (possibly final) point for both XO's. 



> I was focusing on optimizing individual driver SPL using PEQ to obtain a good overall final SPL (TW+MR+WF). I understand you are considering the final SPL as a whole that is to be optimized with all available tools first (XO type, slope, frequency for each HPF & LPF) . And then, SPL is to be optimized globally using PEQ (moderately). Technically, it means we should be using PEQ on Input (all drivers) where I was considering using PEQ on Output (individual driver). Am I right?


You are correct regarding my current preferences and I do prefer input EQ filters. It doesn't impact the phase tracking we achieved. I would still use an output filter if the situation calls for it. 

There are lots of options for the measuring scheme. Good results can be achieved in different ways. It's also very easy to make a mess of it. Your intended action is a good idea. Investigate the various options and select one to implement. I would just point out that those selling speakers to others have no good options except to target the design to a flat 1m target. Those selling room EQ devices tend to use averaged measurements surrounding the LP and target a House Curve. That still leaves numerous options regarding to how to measure and how to select filters.


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## Philippe75

Hello JT,

After several weeks of work and holidays, I found the time to complete my first project. I also used this time to improve my usage of REW and perform several testing. And I must say my family is a bit fed-up with loud sweeps and pink noises in the leaving room...

For instance, I take some times to compare the two SPL optimization ways we discussed earlier: 1) on input or 2) on output. I found that PEQs on output do not have that much impact on phase, and thus sometimes improve it. I also found that PEQ on input were difficult to properly define at XO frequencies as they impact 2 drivers. Therefore, I retain an "intermediate way": limited PEQs on output to optimize SPL for all system usage (TV, Web Radio…) + convolution filter for music thanks to JRiver.

In detail, I did:
1) RAW measurement of individual driver (no filter, no PEQ, nothing)
2) SPL light optimization of individual driver (2 to 5 PEQ maximum)
3) XO definition (Link 36dB at 700 Hz and 8000 Hz, not that much effect on phase indeed)
4) Phase optimization at each XO following the method described in the post
5) Convolution filter derived from phase 4 final result (impulse generated with Rephase)

I can post files if this is of interest to someone (Excel sheets which describe steps 1 to 4 and REW .mdat files).

I am now listening to the result, and I must say I re-discover lots of songs and titles: instruments have a clearer and more detailed sound, some instrument can now be heard (!), the scene is wider, bass are clean...

This is a great improvement of my system!

*Many thanks again for your time, patience, detailed explanations and tips.*


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## jtalden

Thanks for the feedback on your results. It is good to learn that that you have achieved a significant sound improvement. :sn: 

My experience is similar to yours in that there is much more clarity to the individual instruments and the sound stage is improved.

I would be interested to see an .mdat showing the results with your latest EQ settings.

Were you able to measure the results including the impact of the phase correction IR on the system?


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## Philippe75

I will post the .mdat before convolution, i.e. with PEQ and phase alignment at XO freq. But I haven't found yet a way to generate a sweep with REW through the convolution engine of JRiver to measure step 5 benefits. So far, I am using ... my ears.


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## jtalden

That's fine. There is no need for the convolved measurement if it is inconvenient.

FYI:
*HERE* is one thread regarding how to reroute the REW signal so you can measure the impact of the filters applied in JRiver, Foobar, or similar music server. There a probably better threads out there. The detailed info may be on the JRiver forum? I haven't used this method so I can't help with questions about it. Possibly the special routing must be setup for each measurement session if this method is used?

*HERE* is another method. It is the one I use. If you want to use this approach, I can either setup the file for you, or answer questions regarding the steps needed to do it yourself. Once the file is on your music server all you need to do is play it to take your REW measurements.

The file will look like the one shown *HERE*.


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## natehansen66

Philippe75 said:


> I will post the .mdat before convolution, i.e. with PEQ and phase alignment at XO freq. But I haven't found yet a way to generate a sweep with REW through the convolution engine of JRiver to measure step 5 benefits. So far, I am using ... my ears.


You can use the loopback feature in MC to route any audio through MC's DSP. My speakers are active using mc for xo/eq and I use it with measurement sw all the time. I haven't used convolution much but when I've done linear phase correction the measured result appeared to be correct.

http://www.thewelltemperedcomputer.com/SW/Players/JRiver/Loopback.htm

The upcoming version of mc will have its' own wdm driver so this process should be more streamlined and reliable.


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## Philippe75

Thank you very much. I was indeed thinking about kind of "virtual cables" software but I never used one. I will take a look to this feature in coming weeks as I first need to improve 2 or 3 things and especially the convolution filter as it was my first attempt with Rephase.


I will post .MDAT file thiw WE, below are some findings and things I need to analyse:

> Individual driver RAW measurement and SPL optimization was done in one day, having the mic in the exact same position (1m distance, appropriate height) as well as audio path setup. I found that EQ proposed by REW were not giving the same result when live testing. Minor things in REW (Xilica only have 8 EQ per input / output, not 16), but is suppose there are some issues with Xilica PEQ implementation in REW... Therefore, I did the optimization live: manual setup in Xilica and then measurement until I have a good SPL with 1/6 smooth. Time consuming but OK.

> Individual driver measurement (with optimized SPL) for phase optimization and then full range measurement was also done in one day, having the mic in the exact same position (2m distance, ears height at listening position) as well as audio setup. Simulated phase in REW was far from being perfect due to drivers natural phase + Link 36dB XO, but live testing is showing 2 or 3 180° ripples. Did a mistake during phase optimization? Could I done wrong with IR alignment? Xilica delay setup? Measurement technic? There is something I can not explain here... And since I use this last measurement as an input in Rephase, it was a bit tought to have an acceptable phase and SPL as a result.

However, the audio result is already far better than before. I am positive, there are lots of room for final optimization....


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## jtalden

I was mostly interested in looking at the general SPL and phase results at the LP. I wasn't looking for anything beyond a LP measurement of the 2 speakers. In most cases it will not look significantly different when the phase tracking is well tuned. The measured improvement in tracking is not easy to see in an LP speaker measurement. [It's often not that easy to see even in the individual driver measurements!] 

EQ optimization, is an ongoing process for those of us that are in it for the learning experience. There is always more to learn and tweak. As you have found, once we are close with phase tracking and EQ, the sound is noticeable better and continued measurement refinement has little practical effect.

Regarding Rephase:
My approach/recommendation is to use Rephase only to remove the phase rotation of the direct sound - ignoring the phase impact of all the reflections/modes. Depending on the distance of the mic the direct sound phase response is difficult to determine as you have experienced. It just takes some practice - okay, maybe a lot of practice. As you know, it is helpful to look at the phase with the mic closer to the speaker than the LP and to progressively reduce the window length to see the phase of the HF more clearly. That will help identify the true phase rotation for the measurements at the LP. This doesn't help much at LF for those of us using SWs as the direct sound phase really must be measured at the LP. In your case of 3-way mains and no SWs, it should be very helpful there as well. I also often use HolmImpulse to help clarify direct sound phase measurements as it uses a different algorithm when filtering is applied that tends to help clarify the direct sound phase trace. Careful use of trace arithmetic is also helpful. If the FL and FR IRs are manually aligned then the resulting sum (or average) of them may help clarify the phase trace a little. 

Above comments/ideas in case they are helpful.



Philippe75 said:


> > Individual driver RAW measurement and SPL optimization was done in one day, having the mic in the exact same position (1m distance, appropriate height) as well as audio path setup. I found that EQ proposed by REW were not giving the same result when live testing. Minor things in REW (Xilica only have 8 EQ per input / output, not 16), but is suppose there are some issues with Xilica PEQ implementation in REW... Therefore, I did the optimization live: manual setup in Xilica and then measurement until I have a good SPL with 1/6 smooth. Time consuming but OK.


I presume there are 16 filters in REW Xilica so the 8 input and 8 output may be calculated and used together for anyone that wants to do that. To limit the REW calculation to 8 filters just uncheck 8 of the filters in the "EQ Filters" window. That will disable those 8 filters. Selecting "control/manual" for them will do the same. Manual filters are still active, but will not be changed from whatever manual setting they have. There is nothing wrong with manually setting whatever filter settings you feel is appropriate. The REW automated calculation is a convenience.



> > Individual driver measurement (with optimized SPL) for phase optimization and then full range measurement was also done in one day, having the mic in the exact same position (2m distance, ears height at listening position) as well as audio setup. Simulated phase in REW was far from being perfect due to drivers natural phase + Link 36dB XO, but live testing is showing 2 or 3 180° ripples. Did a mistake during phase optimization? Could I done wrong with IR alignment? Xilica delay setup? Measurement technic? There is something I can not explain here... And since I use this last measurement as an input in Rephase, it was a bit tought to have an acceptable phase and SPL as a result.


I do not have a clear understanding the situation, but will make a general comment anyway. The 180° ripples may just be the result of filtering or overly narrow window setting? In that case they may actually be 360° rotations as a result of room reflections. This is a case where HolmImpulse may help to clarify the situation. All rapid phase aberrations are best ignored for phase tracking or Rephase purposes. Only the smoothly changing direct sound phase should be addressed. If your phase trace was as shown in Post 70 for example, then Rephase could be used to remove the direct sound phase rotation as indicated by the dotted red line.

The SPL should not be impacted at all by Rephase if you are following my recommendations. If you are using Rephase differently then I may not understand the logic. I am sure there are many different approaches in use.

If you would like me to look at the measurements and clarify the direct sound tracking or give opinion on other issues I would do that.


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## Philippe75

Of course, you would have corrected by yourself, I meant 180° phase ripples that are more frquent in measured SPL compared to the calculated one. I am not using HolmImpulse, so I need to look at the software first and understand how to use it for a better direct phase understanding and working.

For Rephase, and as you mentionned, I am only using it to adjust Phase measured at 2m distance, not at LP. So I think it is correct. I am not using Paragraphic Gain EQ since individual driver gain and SPL adjustments are done within Xilica.

Below is a .ZIP file containing:
>MDAT with individual speaker measurements (2m distance including SPL optimizations and filter settings but no gain), calculated overall SPL and measured one at same mic position.
>XLSX containing XO descriptions and various information.
View attachment Link 36.zip



Today, I setup the filter with 24 dB slope instead of 36 dB. Phase looks less tormented (less 180° ripples) as well as IR. This phase is easier to work on with Rephase, I now need to compare the 2 filters with my ears....


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## jtalden

Comments on the data:

1. One reason for the difference in your Simulated Vs Measured comparison is that the simulated calculation was done with tight windows still applied to the Supravox and to the Beyma voices. There is only a minor impact to the Beyma SPL, but the Supravox SPL was greatly impacted. For IR math the left and right windows must be sufficiently large. I usually use the default window values as they are very conservative and avoid any risk, i.e., 125, 0, 500, Turkey .25, Turkey .25. The calculated Vs measured should agree very closely when using those default windows.

There also appears to be both a level increase for the Beyma in the measured trace and also different XO SPL support than was calculated; See below. The agreement between calculated and measured will always be perfect in my experience. If you see a significant difference like this level difference in the Beyma or XO support it is a sure indication of a change or problem in the process somewhere. I suspect both the Beyma level was changed for the measurements and the relative IR delays are different for the calculated Vs measured traces.

















2. The "XO" measurements timing has been adjusted and shifted to near 0 ms and I calculate the delays offset entered into the Xilica should therefore be approximately:
Fostex: 1.2 ms
Beyma: 0 ms
Supravox: 0.2 ms

Per the discrepancy in the XO range for SPL mentioned above, I suspect the actual values entered were different from this.

I notice that the Beyma IR is delayed significantly from the other 2 voices. That suggests that the best phase tracking alignment has not been achieved.

3. I decided to go ahead and calculate the delays I would recommend for this setup. These are the only delay settings that provide the closest phase tracking through both XOs. 
Fostex: 1.22 ms - with the *polarity inverted*.
Beyma: 0 ms
Supravox: 1.59 ms

Charts below use this calculated timing:


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## jtalden

I reviewed the data again this morning regarding the 800 Hz XO. I was concerned because the Supravox IR seemed to be trailing more than normal the initial rise of the other voices and the very irregular/noisy phase traces makes the direct phase difficult to accurately determine. Apparently there is significant phase shifting as a result of reflections with that mic position. This is probably what was giving you concern in your initial analysis. [Possibly moving the mic a little closer would have provided a cleaner indication of the direct sound phase by reducing the impact of the reflections? These things need to be experimentally determined. 

In review I found that the above delay setting may not best choice for phase tracking after all. It is probably 1/2 wavelength off. The change needed is to the delay of the Supravox and it also needs to be inverted. The initial recommended timing is still very good so those setting will work fine also. 

The resulting SPL and tracking accuracy is almost identical and there is likely no detectable impact to the sound quality. The only measureable impact will be to reduce the total speaker phase rotation by 180°. The overall group delay will be reduced by 0.69ms. These are both relatively trivial differences.

There was no change to the 8k XO delay setting. 

The corrected delays are:
Fostex: 1.22 ms - with the polarity inverted.
Beyma: 0 ms
Supravox: 0.90ms - with the polarity inverted.

[With these delays an option is to invert the polarity of the Beyma and leave the original polarity of the Fostex and Supravox. There is no practical impact to which option is chosen.]

Charts below use this calculated timing:


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## Philippe75

First, I need to apologize as I now see that the Excel file was not in the ZIP package. You would have seen that delays parameters are as following :
> Fostex (TW) = -1,209 ms
> Beyma (MR) = 0 ms
> Supravox (WF) = -0,210 ms
You did a perfect re-calculation, congratulations!!!!!

In regards of gain, I was not precise enough in my previous post. Indeed, individual speaker measures are with PEQ and XO but no gain adjustment. I only adjusted gain while proceeding with the overall confirmation measure.

In regards of XO, I should maybe first mention that due to speaker shape, TW and WF are in very front of MR because of the horn. This is the reason why I need to add "strong" delays to WF and TW. In that attempt, I did not want to invert impulses. But you are right, this would probably slightly limit phase rotation. I will need to check this also... with a larger IR window. This should be OK now as I can more easily "read" the phase curve even with 180° ripples. Not that fluent, but it is coming.

In regards to mic, I need to see if indeed I would have beter results with a different position in the room. But you found one of the major defaults: my room. And I will hardly have the ability to improve it because of WAF... Jokes put aside, mic positioning is the hardest part of the job, and I can not install speakers outside for farfield measurement.


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## jtalden

Your simulated results on the 8kHz XO were fine. The issue I saw was that the measurement result did not show the SPL support that the simulation did. I am not sure the cause of this, but at 8kHz the wavelength and period are very short so any small timing error is very significant to phase. I was unable to find the resolution of the Xilica delay increments, but only a 0.031ms timing error (1/4 wavelength) would explain the simulated vs. measured results. 

I am working at much lower frequencies and didn't consider the practical issues of this situation. It is very likely that it is just as effective from a sound quality perspective to align the 2 IR's as well as possible with the Xilica increments provided and not be concerned with the resulting phase tracking beyond choosing the polarity that provides the most SPL support.


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## Philippe75

I will recheck measures in REW and settings in Xilica as I may have done a mistake when typing values... For your information, Xilica has a 0,009 ms or more step. If I remember well, TW delay was not the calculated one because of this step factor, but WF was OK.


But, bad news, I will have to change this crossover as I have more and more issues with it, probably related to the PSU (warranty is over). I lost the left voices this WE (was back after several reboots), I have a slight fade-in when I start playing a new song (no specific compressor or limiter settings) and it was totally locked 3 months agao while playing (a hard reset was required).

I don't know yet which new crossover I should select, knowing that RCA (Pre) to Pro XLR (Xilica) to RCA (Amps) audio path is giving me some pain with compression drivers. A way around was to use Jensen transformers to perform Pro XLR to RCA conversion, but I still hear a PSU high frequency background noise... I tested MiniDSP 4x10 Hd (RCA only) but the sound was not as good as Xilica and there was a similar background noise. I also tested a Nova HD (Pro XLR) and had the same noise + an horrible noisy fan.

In addition, if I change the crossover, I would like it to have a better phase management. Therefore, I am considering purchasing a FIR crossover but there are not so many and they are quite expensive. Lake LM26 could be a good choice (but PRO XLR) or DEQX Express 2 (RCA, but also a Pre...). Electrovoice DX46 seems to be limited and AD/DA conversion is 48 kHz only (Xilica 96 kHz).

I am not considering moving to a full PC based solution, it seems to me a bit too tricky or complex to setup.

I would welcome any suggestion. Just hope Xilica will not broke down in the meantime......


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## jtalden

Your comments are consistent with my understanding of the issues and options. I have nothing that is very helpful, but will comment anyway.

JRiver can apparently do it all. I share you reluctance in going that route. That said, I am sure there is help easily available so it is probably a just a matter of working through the issues. I would probably want to learn how to do the setup offline on a second mini system so as not to take my theater out of service while working through it. In my case there are several practical concerns that seem problematic.

I am able to avoid noise issues with the standard DCX unit using 7" cone MR and dome TW, but it requires careful management of gain structure. If your preamp has enough output level capability it may still be possible using your compression drivers. Have you tried using attenuators at the MR and TW amp inputs? I won't be surprised if it is not be possible to eliminate audible noise at the LP in your situation without some mod or additional boxes. The DCX has various mods available for the I/O sections to better match the consumer levels. I am sure you are well aware of that option. It is a pricey option that should work well if you choose good mods. As you indicate, the XO and EQ filters are IIR that way, but so are the voices. A final Rephase filter can remove all the phase rotation as a final step. I tend to believe the final result would be the same as a fully FIR setup if, the IIR phase tracking through the XOs range is realized. 

I'm currently storing and streaming my music at 44.1k (flac). I don't have a concern with 48k processing, but understand it is a hot topic for those that do. I wasn't able to detect any difference via ABX testing using headphones (44.1k Vs 96k) and have seen credible technical information supporting the premise that we shouldn't be able to. That settled it for me. My ears are substandard and I sure wouldn't dispute those that indicate they can. I suspect 96k units also bring along more delay resolution however and since that is a current area of interest and experimentation I wouldn't want to give it up for that reason. The DCX delay increment appears to be 0.00521ms (about 1.8mm) so that well suits my requirements.

I don't have any useful info regarding all the other speaker management options available and carry the same general concerns you expressed.


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## Philippe75

I am in contact with Jimbee, a french hobbyist (http://jimbee.over-blog.com/). He confirmed the Pro XLR / Hifi RCA issue and suggested me to deeply investigate a full PC base solution with external card. Crossovers would be achieved by JRiver thanks to multiple Rephase impulses (one per voice or speaker) and additional delays.

I take a first look at high quality external USB interfaces. Most of these are expensive (but less than a LM26) and have symetrical outputs (TRS or XLR). If this solution looks good (all signal processing is done in digital domain and no A/D conversion), I am still concerned about 3 points:
> If I have issues with Xilica Pro XLR outputs, will I have issues with the soundcard symetrical outputs?
> How to protect amps and speakers in case of bad volume level management? Ideally, a soundcard with a volume knob would be perfect for me and the rest of my family.
> How to manage external sources that are also attached to the Preamp (e.g. TV, TV box HDD recordings, radio)? I need to read posts you mentionned earlier (routing REW to JRiver).

Door is not close so far...


I am possibly wrong, but the gain path doesn't seems to be the problem. For instance, the noise appears when I turn on Xilica and TW amp but no other devices: this simple chain Xilica -> TW amp is the problem. I indeed managed to reduce noise using Jensen PC2XR Pro XLR to conventional RCA converters for MR and TW. They are positionned at the Xilica outputs. The resulting audio path is Xilica XLR -> Jensen XRL to RCA -> RCA amp. If the sound was still there, it was acceptable at LP. But unfortunately, TW is making more noise at LP recently (related to the PSU issue of the Xilica?). I also tested Lundhal Problem solvers and some attenuators (-20 dB), but the best result is achieved with Jensen transformers.

I am indeed aware of DCX mods, but I always thought that, if sound is better, price becomes too high for an adapted device with no warranty...

Like you, I am mostly listening to 44.1 FLAC files converted from my CD tech or purchased on the net. I also have some high definition files but in small volumes. My concern was about the AD/DA frequency within the crossover. Xilica is at 96 and sounds good. As I mentionned, I tested a Nova HD at 192. Noise and fan nois put aside, the overall sound appeared less stressed/digital. But this can come from various differences in crossover components and processing. Difficult to say...


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## jtalden

Just a quick correction: The DCX delay is actually running at the same 96k clock rate, so the increments are 2x the ones stated above or, 0.01042ms (about 3.6mm).

Your decisions on which way to go is not an easy one. It will be a tradeoff in any case. Good luck!


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## Philippe75

The DCX delay is not that far from the Xilica. I suppose this is due to the AD/DA frequency.

So far, I still haven't found how to commonly use one laptop to do everything (i.e. PEQ, filtering, convolution) accepting external signal easily. I'am afraid we are too early in time!!! I won't keep this alternative open.

Strange, this WE the Xilica is working fine (no PSU noise) and this was good as it was before plus the phase optimisation plus the convolution filter. Jensen are working good! This makes me think a new filter should be very nice or a Deqx or an equivalent. Trinnov is out of my possibilities (good product as I heard during fairs but...). I am just concerned Deqx is not distributed in France. This might be a good improvement (all is done in digital domain) but I am not that confident in Deqx........... Nothing pragmatic, just a feeling.

Suspense at its maximum.


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