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Understanding the state of the DSP market

I agree with OCA's opinion.


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I've seen some users export WAV files, then import them as raw files into Audacity and change the file extension to .bin to use them.
I haven't been able to test it myself since I no longer have a MiniDSP.
So, I'm not really sure if that actually works.


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By the way, A few simple examples.
(I still don’t fully understand the API (always grateful to @JohnPM for REW), but users like @OCA automate it and use it effectively for their purposes. - So, if you have a good knowledge of API, you can either automate it or do it manually.)


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.wav to .bin converter:

 
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Wow. You’re a genius. I’ll share your link with the Korean community I follow.
 
It's a good point you made, to contrast crossovers with room correction filters. They don't have anything to do with each other ideally, except perhaps for sub integration.
The comments I made about less FIR taps sounding better, don't really apply to linear-phase crossovers. Even high order crossovers have relatively low Q's, which don't require a lot of taps to implement. What does take a lot of taps are higher Q PEQ's for example, used to smooth out frequency response. Here's where less FIR is more.....which is no surprise I think, as nearly everyone knows high Q IIR filters tend to suck and are overcorrections.

The potential overcorrection with DRC room correction that goes beyond too much frequency response filtering, is if phase is being altered too. I know Dirac uses mixed phase, having run transfer functions on the filters it generates. No clue about others...but generally, ANY time phase linearization is done other than with complementary crossovers, the potential for pre-ring exists.

The other problem with DRC in my mind, is that although each room mode/reflection itself is minimum phase, the summation of any two together is not....much less all of them together.
That summed hodge-podge is what gets corrected with a single filter set. a ...waay too much luck of the measurement draw for me.

DRC simply has to be to a spot at worst, or to a relatively small area at best.
Personally, I ditched the goal of sweet spot listening over twenty years ago...and has proven to be one of the best audio moves I've made in my life ! :D
Well the power of using Acourate or Audiolense is that the filters you create are based on measurements taken at the listening position(s) (which from what I’ve read is a wider area than just one seat, hence the plural in brackets). So if you are creating crossover filters they will automatically be taking the room into account. Ditto for EQing.

I’m certainly not suggesting that this removes the problem of overcorrection (indeed Mitch advocates that less is often more in this context) but this listening position led approach does seem to be the ideal approach in principle as far as I can see.
 
Mixed Phase
This is achieved by cascading an IIR filter with a FIR filter, sometimes the other way around.
great post! my only comment is that mixed phase just means that the filter is in between minimum phase and maximum phase, ie has a minimum phase part and an excess phase part (all pass, delay etc). both IIR and FIR can be mixed phase and minimum phase
 
Well the power of using Acourate or Audiolense is that the filters you create are based on measurements taken at the listening position(s) (which from what I’ve read is a wider area than just one seat, hence the plural in brackets). So if you are creating crossover filters they will automatically be taking the room into account. Ditto for EQing.

I’m certainly not suggesting that this removes the problem of overcorrection (indeed Mitch advocates that less is often more in this context) but this listening position led approach does seem to be the ideal approach in principle as far as I can see.
Not really, acourate particularly is single position based measurements and crossover/speaker design is not room eq
 
Not really, acourate particularly is single position based measurements and crossover/speaker design is not room eq

Acourate can be made to use any curve as the basis of correction, including MMM, multi-point averaged, etc. although the simplest way is to use a single measurement. Uli has explained how to do it before, and the guide I wrote simply replicates Uli's description. The trick is to use one measurement to generate the inversion, then another measurement (single position, which preserves phase and timing information) to generate the correction.
 
Acourate can do pretty much whatever you want at various stages but the actual filter creation part is single point. Anyway the main point is not to conflate this with speaker design, these are two different activities.
 
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