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In Reply to: RE: New "redesign" of LXmini. posted by Davey on October 15, 2016 at 08:01:45
You are saying all PEQ are minimum phase? Everywhere or just the minidsp? If the PEQ in the minidsp are indeed minimum phase, and I personally would not believe that without seeing the source, then that is actually worse. That means the filter implementation is so bad that they sound like linear phase. Either way, please continue...
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PEQ filters can be either, but not so with the basic/cheaper miniDSP units. Only their platforms with more computing power can implement linear-phase filters.
If you don't believe it, a very simple bench test can reveal this behavior.Regardless, (using your over-simplified conclusion) it's actually better not worse...IMO. Minimum-phase PEQ correction filters (applied in this context) actually yield a net linear-phase result, whereas a linear-phase correction filter would yield a non-linear-phase result. Think about that a bit. :)
However, it's arguable that the whole concept of worrying about phase distortion with amplitude corrections is even meaningful.Phase distortion (and the audibility of it) is an interesting side discussion not (directly) related to this. Suffice it to say, there are multiple opinions on it.
Anyways, obviously you fellas can build your design however you like. I don't have any control over that, but what I can/will do is correct mischaracterizations and misinformation regarding the existing LX designs from Linkwitz.
Dave.
Edits: 10/15/16
I probably gave too much credit to MiniDSP in being able to implement FIR with their processing power. Let's just assume I am wrong and they are IIR, which are much more efficient to implement.
As you mention, though, the phase effects from every type of filter are complex. Additionally, different algorithmic implementations have different side effects (usually distortion, but phase and resolution can be affected as well). In theory, correcting the SPL response also corrects phase at the same time. Win win. In reality, the "correctness" of leveling phase correction is probably not as perfect as the theory states and the distortion and other byproducts of having the filter there at all are a real problem.
So, what we observed was a negative effect of having filters in place. We observed them in crossovers, shelf, and peaks. We did see that moving to high precision 32 bit floating minimum phase filters sounded much better (clarity) than whatever the implementation was on the three different MiniDSP products we had. But, by using a driver that did not require any filters (other than a small 2.2k notch) we removed the possibility of introducing distortion or unwanted phase effects and have found that the resulting clarity is far more important than having a ruler flat SPL graph. Additionally, even without a single filter, our ABX tests put the DAC implementation in the MiniDSP dead last against our available test units. The 2x4HD is significantly better then the 2x4 and 4x10, but still not at current audiophile standards. The standard bearer, in my opinion, is the currently unknown dspmusik.
This is actually a perfect illustration of the difference between SLs approach and ours. SL has complete understanding of the theory behind the relationship between SPL and phase and has a mathematically perfect solution to correct both at the same time. He is 100% right that that is the ideal solution and any deviation from that is a step in the wrong direction. The problem is, the implementation of that solution (on the DSP hardware side) is not perfect and the side effects of attempting the solution are audible and problematic.
So, what do you do? You could wait for the DSP implementation to catch up with the theory and eventually have a correct solution that sounds perfect. Pros: You are proven right. Cons: May take a while to get there. In the meantime, you are getting your ass kicked in clarity by KEF and other manufacturers that take a more pragmatic approach. Or, you could go back to the fundamental nature of the problem, which is the use of a driver that requires so much PEQ in the first place, and redesign a system that does not rely so heavily on filters. Pros: You might be able to make something subjectively sound better. Cons: Everyone thinks you don't understand the theory and therefor your solution must be wrong.
The thing is, we do not have the same design goals as SL. He is making a transducer that converts electrical impulses to sound waves. We are making a speaker that sounds as good as possible given the limitations of driver construction, analog and digital filters, and all the other factors that go into speaker design. A lot of times this means having a solid grasp of the theory and then being willing to throw that out the window and try a bunch of stupid shit until you find something that works better. As my dad likes to say, If it's stupid, and it works, it isn't stupid.
So, throw the theory out the window and try a bunch of stupid shit until it sounds/works better? I'm glad your design acumen is not squishy or anything. :)
Take care.
Dave.
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