I don't think an FIR is emulating an IIR as such, they are just 2 different types of filter with their own sets of pros and cons. The minidsp site has a basic overview, a slightly more in depth one at dspguru (and obviously you can google iir vs fir to your hearts content ).
Sorry, I think I've confused things, and I probably shouldn't have used the word "emulation." What I was trying to get at is that, as I understand it, performing minimum phase EQ in a FIR filter will not be as efficient as performing the identical minimum phase EQ using an IIR filter:
The main advantage digital IIR filters have over FIR filters is their efficiency in implementation, in order to meet a specification in terms of passband, stopband, ripple, and/or roll-off. Such a set of specifications can be accomplished with a lower order [...] IIR filter than would be required for an FIR filter meeting the same requirements. If implemented in a signal processor, this implies a correspondingly fewer number of calculations per time step; the computational savings is often of a rather large factor.
[from the IIR wiki page]
I didn't mean that it was literally attempting to emulate the exact processing steps, more that it was attempting to create the same minimum phase result, but that it required more time and computational power to accomplish it (in the same way that a NES emulator requires much better hardware to run than the NES it's emulating).
I would think that the low latency filters produced by acourate/audiolense are just minimum phase FIR filters, perhaps with fewer taps to boot. This means no phase correction & potentially less accurate frequency response correction but lower latency and perhaps a good enough amplitude correction.
I think we're agreeing on this part. I assume acourate and audiolense are using FIR filters to create a minimum phase response; I'm not sure how they could be doing anything else when their only output is a FIR filter. They definitely would have to reduce the taps to reduce the overall latency. The part that I was expressing bafflement about above is that, to my understanding, there are certain minimum latencies involved in doing any kind of FIR manipulation (even manipulation of frequency response), and that those minimum latencies increase as the frequency of interest falls. So what I'm not quite sure about is how they manage to do any meaningful frequency correction below, say, 60 Hz with a 10 or 15ms FIR filter. Maybe the answer is, as you say, it's "good enough" correction, or maybe the limitations of FIR latency are not where I think they are
The distinction I was making about Dirac is that it's not necessarily limited to using only FIR filters, because they're running their own audio driver. So they could have an IIR stage, and then an FIR stage and get the best of both worlds in terms of latency and flexibility. But to be clear, I have no idea what Dirac actually does. Their literature just says their method is not fully IIR or FIR, and so I made an inference.
It might be interesting to see some comparisons of the phase effects in mojave's earlier charts.
100% agreed, I would expect to see a much greater divergence there.