^2.83V is nice voltage if measurement system is SPL calibrated and measurement distance is 1m. For example I measure hifi drivers with 2.83V if crossover will be passive. 2V is a bit safer but needs +3dB scaling in Drivers tab to give USPL with calibrated gear. Compression drivers can be measured e.g. with 0.5 V to reduce SPL below 100 dB/1m.
So there is no single globally right measurement voltage. 2V was for your Revelator and gear without SPL calibration.
So there is no single globally right measurement voltage. 2V was for your Revelator and gear without SPL calibration.
Sure I will need time to look and experiment with it.
Rev 2.0.5.6 (2018-11-04)
Enclosure tool
* Driver database filter extended to cover all fields. Filtering criteria moved to separate window which opens with Filter button. Numerical parameters are filtered with min and max text boxes or vertical sliders. Added indication of visible and total rows or 'wait...' while filtering.
* Added parameter cross calculation to enclosure with passive radiator: PR.Vas is calculated if PR.Sd, PR.Mms or PR.fs is changed. PR.fs is calculated if PR.Vas is changed.
* Added read-only text box fe[Hz] 'fs with extra mass' to enclosure with passive radiator.
Enclosure tool
* Driver database filter extended to cover all fields. Filtering criteria moved to separate window which opens with Filter button. Numerical parameters are filtered with min and max text boxes or vertical sliders. Added indication of visible and total rows or 'wait...' while filtering.
* Added parameter cross calculation to enclosure with passive radiator: PR.Vas is calculated if PR.Sd, PR.Mms or PR.fs is changed. PR.fs is calculated if PR.Vas is changed.
* Added read-only text box fe[Hz] 'fs with extra mass' to enclosure with passive radiator.
I have to say again that the ore I use it, the more I am excited with Kimmo and Vituixcad! Even if a few more people were so good in everything they do, our world would be much better for everyone!
Following some ideas that would be great if could be included in new updates.
. When we add active filters to the passive filter, the impedance disappears. However, it is important to be visible and remain within minimum limit, when we run the optimizer including both passive and active filters. Is it possible?
2. It is important to know if a filter peak comes from the active equalization or from of a passive filter. It is important to use the gain limit of the optimizer only for the active EQ, because such a high peak reduces the amplifier output in the other frequency areas.
Could we have a choice of limiting only the active gains or both passive and active gains?
3. Could we choose to apply max limit only for selected drivers? Or selected frequency areas?
4. It seems even more interesting to put different gain limits per driver or for certain frequency areas.
5. Can we have an optimizer also for Directivity index? This is very important for the sound quality.
6. Can we allow also the polarity of the drivers as parameters for the optimizer? It is helpful if we have more than 3 ways and if we want to optimize the Directivity index.
Following some ideas that would be great if could be included in new updates.
. When we add active filters to the passive filter, the impedance disappears. However, it is important to be visible and remain within minimum limit, when we run the optimizer including both passive and active filters. Is it possible?
2. It is important to know if a filter peak comes from the active equalization or from of a passive filter. It is important to use the gain limit of the optimizer only for the active EQ, because such a high peak reduces the amplifier output in the other frequency areas.
Could we have a choice of limiting only the active gains or both passive and active gains?
3. Could we choose to apply max limit only for selected drivers? Or selected frequency areas?
4. It seems even more interesting to put different gain limits per driver or for certain frequency areas.
5. Can we have an optimizer also for Directivity index? This is very important for the sound quality.
6. Can we allow also the polarity of the drivers as parameters for the optimizer? It is helpful if we have more than 3 ways and if we want to optimize the Directivity index.
...
About you loose impedance curves think can be overcome if you put in a OPA or power amp (the two triangle symbols) into chain in strategic right places, as example see graph post 691, that is pure active model but still it show impedance because triangle symbol poweramp teminate that chain before speaker driver, if i then right click poweramp and thick short then impedance curves will miss although all else will stay the same.
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Many thanks Vassilists for feedback!
I didn't read diyaudio first so questions were answered by e-mail, but here is short version for the others.
1) Active buffer block is capable to measure load impedance. Add active buffer (=power amplifier) to the last block of each "way" to show impedance curve.
2) Manual checking of gain with "probe driver" ie extra driver which is electrically open is the easiest method to check gain in any point of filter chain. I suppose that signal is also possible to optimize by gain limit but it's not easy enough for daily use.
3...4) See item 2. I don't like to make optimizer too complex. It should be simple enough to be overall faster than manual adjustment, and attractive for users.
5) DI is relation of axial and power so it will be optimized automatically (as much as possible) when axial and power are optimized with about 50/50 weighting.
6) Polarity change would make big jump to error which will be rejected. Better to save two different scenarios and compare which one is better in the end.
I didn't read diyaudio first so questions were answered by e-mail, but here is short version for the others.
1) Active buffer block is capable to measure load impedance. Add active buffer (=power amplifier) to the last block of each "way" to show impedance curve.
2) Manual checking of gain with "probe driver" ie extra driver which is electrically open is the easiest method to check gain in any point of filter chain. I suppose that signal is also possible to optimize by gain limit but it's not easy enough for daily use.
3...4) See item 2. I don't like to make optimizer too complex. It should be simple enough to be overall faster than manual adjustment, and attractive for users.
5) DI is relation of axial and power so it will be optimized automatically (as much as possible) when axial and power are optimized with about 50/50 weighting.
6) Polarity change would make big jump to error which will be rejected. Better to save two different scenarios and compare which one is better in the end.
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Rev 2.0.5.7 (2018-11-09)
* Added listening window average response calculation. Average response is shown in SPL and Power&DI charts in yellowgreen. Curve is possible to save or load to overlay in SPL chart.
* Added Listening window hor± and ver± text boxes to Options window. Initial values are hor ±30 deg and ver ±5 deg. Available range is 0-180 deg for both.
* Added 'Listening window average as axial response' checkbox to Optimizer. If checked, Optimizer uses average instead of single response to reference angle while optimizing with Power response. Target line in SPL chart should be adjusted for average.
Convert IR to FR
* Added 'Full resolution' checkbox. Exports unsmoothed high resolution frequency responses. Frequency step is constant sample rate / FFT length allowing long excess delay without wrapping errors in phase response. Warning: Use short FFT to avoid huge file size and slow processing.
2.0.5.6 (2018-11-07)
Impulse response
* Added sample rates 88200 and 176400 Hz.
* Minimum Taps decreased from 512 to 256.
* Taps text box changed to wheeling. Few other cosmetic changes.
* Added listening window average response calculation. Average response is shown in SPL and Power&DI charts in yellowgreen. Curve is possible to save or load to overlay in SPL chart.
* Added Listening window hor± and ver± text boxes to Options window. Initial values are hor ±30 deg and ver ±5 deg. Available range is 0-180 deg for both.
* Added 'Listening window average as axial response' checkbox to Optimizer. If checked, Optimizer uses average instead of single response to reference angle while optimizing with Power response. Target line in SPL chart should be adjusted for average.
Convert IR to FR
* Added 'Full resolution' checkbox. Exports unsmoothed high resolution frequency responses. Frequency step is constant sample rate / FFT length allowing long excess delay without wrapping errors in phase response. Warning: Use short FFT to avoid huge file size and slow processing.
2.0.5.6 (2018-11-07)
Impulse response
* Added sample rates 88200 and 176400 Hz.
* Minimum Taps decreased from 512 to 256.
* Taps text box changed to wheeling. Few other cosmetic changes.
Thanks for all the updates as always.
Since there are FIR filter blocks, I think it would be awesome to have some convolution possibilities.
So let's say use a reference response (either target or another system) and take the inverse difference compared to the system you're designing.
That way it's very easy to build a filter according to a set of requirements.
One other thing is if it would be possible to work with relative folders and files?
A friend of my send me a project (plus folders) to have a look at.
VituixCAD is still looking (and showing) the old directories, so it basically means I have to add everything manually again.
Since there are FIR filter blocks, I think it would be awesome to have some convolution possibilities.
So let's say use a reference response (either target or another system) and take the inverse difference compared to the system you're designing.
That way it's very easy to build a filter according to a set of requirements.
One other thing is if it would be possible to work with relative folders and files?
A friend of my send me a project (plus folders) to have a look at.
VituixCAD is still looking (and showing) the old directories, so it basically means I have to add everything manually again.
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Hi b_force,
No guarantee for sound result so be carefull with volume but think this should work:
1. File/export/Frequency response
2. Tools/load previous file into A/pick: Mirror A and "Calculate & Save"
3. Open a new or second session of VituixCAD
4. In XO tab give default driver a strait wire and load that previous response to default driver and you see your point 1 response in reverse : )
5. Into View/Impulse Response window its possible now output you a convolution with parameters that suit your situation.
Think whatever response we have as frd or txt file can be translated to a convolution file as kimmosto kindly guided into post 679 and is what happen in above guide from point 3-5, so what happens into point 1-2 above is up to you and most responses can magic be reversed one or the other way in VituixCAD "Calculator" or REW "SPL Tab", another way is for example if you divide a curve with itself and then divide that flat sum with the first curve you get a reverse curve that can be tranlated to IR (point 3-5).
Should tails in txt or frd file be wrong then use Tools/Calculator/Minimum phase A/Response tails to udjust tails and if final IR is not normalized then use free Audacity to correct this.
No guarantee for sound result so be carefull with volume but think this should work:
1. File/export/Frequency response
2. Tools/load previous file into A/pick: Mirror A and "Calculate & Save"
3. Open a new or second session of VituixCAD
4. In XO tab give default driver a strait wire and load that previous response to default driver and you see your point 1 response in reverse : )
5. Into View/Impulse Response window its possible now output you a convolution with parameters that suit your situation.
Think whatever response we have as frd or txt file can be translated to a convolution file as kimmosto kindly guided into post 679 and is what happen in above guide from point 3-5, so what happens into point 1-2 above is up to you and most responses can magic be reversed one or the other way in VituixCAD "Calculator" or REW "SPL Tab", another way is for example if you divide a curve with itself and then divide that flat sum with the first curve you get a reverse curve that can be tranlated to IR (point 3-5).
Should tails in txt or frd file be wrong then use Tools/Calculator/Minimum phase A/Response tails to udjust tails and if final IR is not normalized then use free Audacity to correct this.
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Paths are relative in main project file, but requirement is that response data is in the same directory with vxp or any subdirectory below that. Then you can make backup or transfer to cloud or to friend etc. e.g. with 7-zip batch located in VituixCAD\Projects folder. Parameter %1 is project folder.
Impulse response window converts filter gain of selected driver to IR. Mirroring or other EQ tricks are not needed. Just select drivers one by one, export IR and load to convolver. Takes few seconds but it's not the same as integrated real time convolver. Much better than nothing.
Code:
@ECHO OFF
IF .%1.==.. GOTO EXIT
"%programfiles%\7-Zip\7z.exe" a -r "%1.zip" "%1\*.vx*" "%1\*.txt" "%1\*.frd" "%1\*.zma"
:EXITImpulse response window converts filter gain of selected driver to IR. Mirroring or other EQ tricks are not needed. Just select drivers one by one, export IR and load to convolver. Takes few seconds but it's not the same as integrated real time convolver. Much better than nothing.
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Hi Kimmo,
Your software has come along beautifully.
Quick suggestion on the Calculator. When adjusting tails for Minimum Phase calculation, you might want to retain the actual measurement while adjusting tails so that you can fine tune the tail while looking at the actual response prior to MP calculation.
Thanks again,
Jay
Your software has come along beautifully.
Quick suggestion on the Calculator. When adjusting tails for Minimum Phase calculation, you might want to retain the actual measurement while adjusting tails so that you can fine tune the tail while looking at the actual response prior to MP calculation.
Thanks again,
Jay
This depends on crossover frequency and size of cone, but usually far field measurements are still needed for phase/timing matching with mid-range and tweeter (if measurements are not processed minimum phase calculation) and for directivity information to low frequencies. Also summing of cone(s) and port(s) is easiest with Merger tool which can easily include baffle effect response to near field measurement(s), and merge previous near field total to far field with directivity data in off-axis responses.
Merger youtube lesson shows this process.
That would be helpful, but overlay of Scale function before Minimum phase function makes the same visual effect, though it's slow method if you need to check the result with several off-axis responses before Save command.
Personally I don't like to produce measurement data by cropping, extending and minimum phase extraction to whole frequency range, without applying measured phase to higher frequencies. But this function could be useful to create low frequency part for merging with high frequency part with measured phase.
Rev. 2.0.5.8 (2018-11-11)
* Added Part # text box to Crossover tab for manual editing. Part numbers do not change automatically when schematic is changed.
* Added semi-automatic part renumbering. New commands in context menu of crossover schematic: Number blanks, Renumber all 123..., Clear selected, Clear all. Part # text box is green while numbering mode is on. Numbering is done by clicking components in desired order. Numbering mode ends by clicking empty area or right click or Esc.
* Added Comment text schematic object. T button in component menu or T key. Text can be rotated.
* Added Part # text box to Crossover tab for manual editing. Part numbers do not change automatically when schematic is changed.
* Added semi-automatic part renumbering. New commands in context menu of crossover schematic: Number blanks, Renumber all 123..., Clear selected, Clear all. Part # text box is green while numbering mode is on. Numbering is done by clicking components in desired order. Numbering mode ends by clicking empty area or right click or Esc.
* Added Comment text schematic object. T button in component menu or T key. Text can be rotated.
Not sure that I agree with you. The Scale function doesn't really help you establish an angle of tail from what I can tell. Overlapping this (like in Soundeasy), helps you to estimate a more accurate tail. I know that you prefer the automatic method using the last octave, but this would be another option for those of us used to the SE method. Thanks again for this great software and support.
Red is overlay with Scale function and blue is result, cropped below 500 Hz and extended 24 dB/oct with minimum phase function.
Original measurement is in red curve - as much as data exists. Possible missing tails are extrapolated by default but that does not change what you can do for them with MP function of Calculator.
An externally hosted image should be here but it was not working when we last tested it.
I thought this is what you meant.
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