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Thread: How to set lpads when using DSP?

  1. #16
    Senior Member pos's Avatar
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    Wow that is some serious subs right there!
    I think you should probably cross them lower than 80Hz: the 2235 are capable woofers!

    Regarding the allpass filter thingy, in fact you would be better off with plain filters, and it will be easier to explain:
    So imagine you want to simulate a cascading filter like in Linkwitz' drawing: the signal is filtered for the sub and the high-passed signal goes to the woofer, which in turn passes the high-passed signal to the mid...
    As you cannot do such cascaded filters with your miniDSP you just have to "replicate" the filter setting of the lower channels on each channel.
    So for example if you woofer has a 24dB/oct acoustical HP filter at 80Hz then you just have to dial an electrical 24dB/oct HP filter for your mid...

    Regarding the sweep, yes make them start as low as possible for the driver you are testing. So that would be something like 10Hz for the subs, and maybe 50Hz for the mid...
    No need to push the volume, especially with a close range measurement: you should not have excursion problems.

    Do post those measurements on the forum in txt or wav format (do not forget to plug the vents on the 4343), I will try to simulate some EQ and filter settings.

  2. #17
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    Post of proudness

    pos, they are excellent drivers, high linear excursion, xmax = 33.7mm and xmech = 44mm with the big enclosure the system is quite efficient too. I've pumped about 70V into each (measured at the binding post) and the maximum power draw at the socket has been only 1381w for both(!) when driving them really hard with music. They have continous power handling of 2kW and are made to be excellent in small enclosures of 100L and even less. It's important with some space behind the pole vent for airflow. You can read and see measurments of this driver in 100L closed box @ data-bass.com if you're interested.

    This is what I've been spending my past 4-6 months on, estimating in WinISD, learning about building materials, tools and closed designs. They are well braced, but oscillate a little bit when I run 10 Hz bursts @ 70V due to the share force. They sound really good too. Not like a max spl type woofer. Problem is they're expensive, and I've strained my budget hard by getting them. Started with one in 104L, but ended up with two of these with about 155L net internal working volume to get more efficiency and less distortion due to compression (not that it was an issue with the 104L design).

    Have used plenty of EQ on the low end through close range measurements and linkwitz-transform through biquad in MiniDSP. This was my first venture into DIY, I'm very proud but have not done any finish work as I'm in the experimental phase.

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  3. #18
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    Quote Originally Posted by pos View Post
    I think you should probably cross them lower than 80Hz: the 2235 are capable woofers!

    Regarding the allpass filter thingy, in fact you would be better off with plain filters, and it will be easier to explain:
    So imagine you want to simulate a cascading filter like in Linkwitz' drawing: the signal is filtered for the sub and the high-passed signal goes to the woofer, which in turn passes the high-passed signal to the mid...
    As you cannot do such cascaded filters with your miniDSP you just have to "replicate" the filter setting of the lower channels on each channel.
    So for example if you woofer has a 24dB/oct acoustical HP filter at 80Hz then you just have to dial an electrical 24dB/oct HP filter for your mid...

    Regarding the sweep, yes make them start as low as possible for the driver you are testing. So that would be something like 10Hz for the subs, and maybe 50Hz for the mid...
    No need to push the volume, especially with a close range measurement: you should not have excursion problems.

    Do post those measurements on the forum in txt or wav format (do not forget to plug the vents on the 4343), I will try to simulate some EQ and filter settings.
    The 2235h are excellent. I absolutely love how they sound (which is why I've been wondering about 4355! My brother has a very large listening space and might end up getting them (he loves my 4343B a lot and hardly talks about anything else, hehe), but for my current room they're to big

    I've experimented with different crossovers as these Ultra woofers can actually be crossed quite high too. As we know, my xover settings has not been perfect, so I will continue to experiment with different settings within the 4x10HD. With miniDSP we can have up to four different configurations live in the device at any given time, and you can also import settings. It's just a wonderful device. More advanced filters can be implemented through biquads, see:

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    One biquad for the HPF and one for the LPF. (This screenshot is not from any of my active filters).

    miniDSP has this huge spreadsheet out to find the coefficients for different filters (link to spreadsheet), I've used it to find the coefficients for the Linkwitz-Transform on the subs. Maybe I can use that to construct something more advanced.

    Thank you, I'll try to do it the simple way w/o all pass filters first then, and try to improve upon that when I've found something which works well.

    The reason why I'm using two measurment setups (UMIK-1 and all of what I showed you earlier with the diagram) is because REW does not allow for time coherent measurements when using USB microphones such as UMIK-1. To get proper timing info in REW, you need to have a loopback signal to discount the latency through the signal chain. As it stands, this is not possible with REW w/USB devices. I'm hearing Holm has capabilities for proper timing with USB so I should probably give it a go, as the dB-meter and everything is really complex in comparison to the excellent UMIK-1.

    I can put the raw data from everything on Google Drive and link it from here if you or anyone else would like to look. Can also do txt and wav, np. Will show room response later too, you guys will probably laugh, because it's far from linear, but I like it.

    Also, I'm hearing that the lead man on the Equalizer APO open source project has said that they will incorporate delays and more into the software, so it will probably turn into a super flexible DSP solution for those who are willing to build something with many audio I/O.

    Unfortunately, I didn't have much time to work on the next version of E-APO
    recently, but I will get back to it soon. This next version will contain a Delay
    command and it will also be possible to copy audio to additional "virtual"
    channels so that you can apply processing to them and mix them back later.

    So you might not be able to directly adjust the delay
    of frequency ranges but with some lines in the config file you can split the
    audio into virtual channels for specific frequency ranges, apply the delay to
    them and then add them together to form the output. You can ask me for an
    example of that when the next version is released if you want.

    Phase correction will not be possible yet, however.
    So no phase correction soon, but these are exciting times considering the computational power of computers nowadays.

  4. #19
    Senior Member pos's Avatar
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    Very impressive construction, I bet it sounds fantastic.
    I wish I had your woodworking skills

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