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Multi-Way Conventional loudspeakers with crossovers

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Old 9th March 2010, 11:05 PM   #51
Theo404 is offline Theo404  United Kingdom
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Quote:
Originally Posted by StigErik View Post
Your right - PLParEQ is very CPU heavy, and the latency is loooooong. Price.. well, it depends. The full version is 1000$, not cheap, but I think its worth the money. More value than expensive cables at least.....

I have a 1.8 GHz CoreDuo machine, and running 4-way XO with EQ on this is difficult, I'm balancing on the edge so to speak, with 97% or so CPU util.....
Are you kidding? $1000 for an incredibly simple piece of software (I use the term lightly, a plugin for a host program is hardly a functional piece of software)? And its not even efficient?!

You people seriously need to consider linux... I run 3 way mains plus a horn sub with long (65536 tap, read: detailed) filters on an ancient athlon system and I dont even approach half CPU capacity. This includes crossover, eq and phase correction for every driver... thats 21 filters. The idea that you cant even run crossovers and eq on a modern core2 system with this awful "plugin" baffles me....

I assume you have managed to fudge around the windows kmixer? Your setup is crippled if you haven’t.

You spoke about driver issues, adding random delay to channels because it assumes you have 5.1... another example of why using windows for audio isn’t a good idea... Alsa will give you direct access to the hardware on your card, no layers of idiotic "value adding" software between you and the outputs.

You run the stream in through an A/D converter before you apply your filters? Why? I can only assume your using a CD "transport" with all of the errors that introduces. Why would any sane person decode digital data from an optical media with their expensive transport "correcting" the read errors on the fly to a D/A converter in it, only to then pass it through a A/D, apply filters and then out through another D/A...? I cant gleam from your posts, but you may actually be putting out SPDIF from your player and taking it into the PC that way... but even this is inherently inferior to direct playback from the PC. Which is another thing my ancient athlon system (I'm talking socket A K7, not 64) can do at the same as all the detailed FIR filters... decode lossles compression formats into PCM streams.

As for the OPs system... still seems like a work in progress. You talk about flat off axis response and then claim large improvements from tilting the drivers so your listening on axis... something that could only narrow the sweet spot and require different eq. If its a matter of time alignment then this can be done on the PC. Seems like your still chasing your tail.
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Old 9th March 2010, 11:14 PM   #52
sqlkev is offline sqlkev  United States
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what is your propose set up when it comes to PCXO? (front end softwares/vst plugins..etc)
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Old 9th March 2010, 11:27 PM   #53
Theo404 is offline Theo404  United Kingdom
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Originally Posted by sqlkev View Post
what is your propose set up when it comes to PCXO? (front end softwares/vst plugins..etc)
A linux system with MPD (music player daemon) piped to brutefir. MPD is controllable with a plethora of other software, directly or over a network..

Brutefir is an incredibly efficient way of applying FIR filters, and has great flexibilty in the structure of them you create. You can cascade as many filters as you like, or combine them all into one. Generation of these filters can be done in many ways, matlab/octave, numerous freeware programs, holm impulse....

Eq and phase correction filters, I feel, can be best created using holm impulse, the many ways to export measured data mean it can be manipulated in any way you like to create detailed accurate filters.
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Old 9th March 2010, 11:42 PM   #54
cuibono is offline cuibono  United States
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Quote:
Originally Posted by Theo404 View Post
Are you kidding? $1000 for an incredibly simple piece of software (I use the term lightly, a plugin for a host program is hardly a functional piece of software)? And its not even efficient?!

You people seriously need to consider linux... I run 3 way mains plus a horn sub with long (65536 tap, read: detailed) filters on an ancient athlon system and I dont even approach half CPU capacity. This includes crossover, eq and phase correction for every driver... thats 21 filters. The idea that you cant even run crossovers and eq on a modern core2 system with this awful "plugin" baffles me....

I assume you have managed to fudge around the windows kmixer? Your setup is crippled if you haven’t.

You spoke about driver issues, adding random delay to channels because it assumes you have 5.1... another example of why using windows for audio isn’t a good idea... Alsa will give you direct access to the hardware on your card, no layers of idiotic "value adding" software between you and the outputs.

You run the stream in through an A/D converter before you apply your filters? Why? I can only assume your using a CD "transport" with all of the errors that introduces. Why would any sane person decode digital data from an optical media with their expensive transport "correcting" the read errors on the fly to a D/A converter in it, only to then pass it through a A/D, apply filters and then out through another D/A...? I cant gleam from your posts, but you may actually be putting out SPDIF from your player and taking it into the PC that way... but even this is inherently inferior to direct playback from the PC. Which is another thing my ancient athlon system (I'm talking socket A K7, not 64) can do at the same as all the detailed FIR filters... decode lossles compression formats into PCM streams.

As for the OPs system... still seems like a work in progress. You talk about flat off axis response and then claim large improvements from tilting the drivers so your listening on axis... something that could only narrow the sweet spot and require different eq. If its a matter of time alignment then this can be done on the PC. Seems like your still chasing your tail.
How about rather than whining, starting your own thread detailing what you do. Recognize that not everyone wants to spend their time programing computers and operating systems.

WRT my system, did you see the title? Doesn't "evolution" imply 'work in progress'? Doesn't diyAudio imply "work in progress"? Changing the axis of radiation is not a matter of time alignment - if you don't understand what we are doing here, you shouldn't be so quick to criticize. I think the civil thing would be to ask why we are doing what we are doing, rather than cast off a bunch of unfounded criticism.
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Old 9th March 2010, 11:55 PM   #55
Theo404 is offline Theo404  United Kingdom
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Originally Posted by cuibono View Post
There are two major differences between the two arrangements: first, each driver's AC is equidistant to the ear; second, each driver's axis is pointed directly at the ear. I am fairly certain that both of these conditions are necessary to ensure proper acoustic summing and power balance between all drivers.
Sounds like time alignment to me....

Apologies for the long post that was slightly off topic (you do have DSP in your title though)... I was more reacting to the idea that anyone would spend $1000 on something so pointless... at least cables are physical.

Of course all of our systems here at diyaudio are works in progress... with my last paragraph I may have been suggesting you may need to re-examine your eq after changing the position of the drivers.
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Old 10th March 2010, 12:13 AM   #56
cuibono is offline cuibono  United States
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First, time physical time alignment has different results than time delayed alignment. The sort of alignment I do is not possible using a time delay. You can read up on this elsewhere.

Second, I did remeasure my EQ - "for this trick to work, each driver must be measured directly on the driver's axis".

Third, tilting the drivers is a mater of balancing the direct response versus total power radiation, and is not related to time alignment.

Fourth, don't assume, without trying it yourself, that the adjustments I've made "could only narrow the sweet spot". Assumptions are a lot more crippling than a slow computer or a long latency.

I'm still interested in other solutions to PCXO, as I'm sure others are. If you started a detailed thread describing what your system comprises of, I'm sure people would appreciate it. Personally, I've always avoided linux just because I have enough to deal with already - if you could make it simpler for people like me, that would be great.
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Old 10th March 2010, 12:43 AM   #57
Theo404 is offline Theo404  United Kingdom
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First, time physical time alignment has different results than time delayed alignment. The sort of alignment I do is not possible using a time delay. You can read up on this elsewhere.
Do you believe your modification gives rise to a different dispersion pattern? I see no mention of this in previous posts and it seems the only positive result possible from your change that couldnt be achieved with delay.


Quote:
Second, I did remeasure my EQ - "for this trick to work, each driver must be measured directly on the driver's axis".
I disagree with this statement. If you flatten fr on axis you are not correcting the power response. On axis only represents a fraction of the power response.


Quote:
I'm still interested in other solutions to PCXO, as I'm sure others are. If you started a detailed thread describing what your system comprises of, I'm sure people would appreciate it. Personally, I've always avoided linux just because I have enough to deal with already - if you could make it simpler for people like me, that would be great.
It distresses me slightly that this is the opinion of many people in the DIY scene.... you can spend countless hours reading up on theory and design, but because computers are 'newfangled' anything but the obvious (windows) is too much effort, yet going out of your comfort zone with acoustic ideas and principles is common place. This is especially annoying when it seems obvious to me that ALL audio reproduction will be computer based in the not too distant future.
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Old 10th March 2010, 04:44 AM   #58
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It would be great to get some advice from the experts about the following things I'm hoping to do. This seems like a good thread to ask these questions.

I have a semi-omni directional loudspeaker that I'm intending to take into an anechoic chamber, make some measurements and use FIR filters to adjust the frequency response of the speakers.

As I understand it, if I put the microphone into the sweet spot (1m away, directly in front, on axis), measure the frequency response of the speakers, and then make that response flat in the frequency domain using FIR filters then there should be the following result: the on axis response will be flatter and because it's in an anechoic environment (minimum phase) the impulse response will be closer to an ideal response, with less ringing in the time domain.

That's the good news. However I have read that the bad news is what happens to the off axis response. That is, improving the on axis response necessarily degrades the off axis response.

I was thinking about this and wondering whether it's actually likely to be true. For instance, let's say that the speaker’s off axis response is quite close to the on axis response. That's the sort of response I associate with the term “constant directivity” speakers: as the angle of measurement moves from 0° in front of the loudspeaker to wider and wider angles, the frequency response measurement tracks fairly well with the 0° response.
If that's so then wouldn't it be the case that as we improve the frequency response of the 0° measurement we also by direct association improve the frequency response of the off axis response?

The second question is: so you have a loudspeaker where the on axis response is a certain shape and the off axis response is actually quite different from that. If you improve the on axis response and therefore cause a change to the off axis response, given that the off axis response is probably pretty bad, and probably un-correlated with the on axis response, is it plausible that the off axis response could in fact be improved instead of degraded, and that there is no way of knowing in advance until you actually do a listening test or off axis measurement?

Hope this all makes sense. If any knowledgeable people can shed light on these questions it would be great to hear from them. Apologies in advance if these seem naive questions to be asking.
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Old 10th March 2010, 07:42 AM   #59
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Quote:
Originally Posted by Theo404 View Post
Are you kidding? $1000 for an incredibly simple piece of software (I use the term lightly, a plugin for a host program is hardly a functional piece of software)? And its not even efficient?!

You people seriously need to consider linux... I run 3 way mains plus a horn sub with long (65536 tap, read: detailed) filters on an ancient athlon system and I dont even approach half CPU capacity. This includes crossover, eq and phase correction for every driver... thats 21 filters. The idea that you cant even run crossovers and eq on a modern core2 system with this awful "plugin" baffles me....

I assume you have managed to fudge around the windows kmixer? Your setup is crippled if you haven’t.

You spoke about driver issues, adding random delay to channels because it assumes you have 5.1... another example of why using windows for audio isn’t a good idea... Alsa will give you direct access to the hardware on your card, no layers of idiotic "value adding" software between you and the outputs.

You run the stream in through an A/D converter before you apply your filters? Why? .
Hi-resolution Phase-Linear EQ will never be efficient, but I don't care as long as it works for me. If the computer runs at 15% or 95% CPU load really doesnt matter I think. Of course - convolver type of filters are much more efficient. The same goes for (phase-warping) IIR digital EQ plugins.

There are spesific reasons that PLParEQ uses a lot of CPU. One of them is that it upsamples the data before processing.

I can only answer for myself here: I use a soundcard (RME Fireface 800) that bypasses the kmixer. I also use a pure digital source (Squeezebox), so there is no A/D conversion involved either.
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Old 10th March 2010, 07:47 AM   #60
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Originally Posted by boconnor View Post
However I have read that the bad news is what happens to the off axis response. That is, improving the on axis response necessarily degrades the off axis response.
That is why Constant Directivity is important. Frequency response should be the same both on-axis and off-axis. It is possible to do that over a large frequency range. It will require that all drivers are operated below the dipole peak.
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