How to use FIR filters without causing delay

Thank you for the great video! I'm just getting started with FIR filters, and at this point my questions are little..basic. It seems like you applied that filter to almost the entire response of the system...so I assume the there is already a crossover LF to HF? Is that the way FIR filters are customarily used? Or can you use them to create the Lowpass/highpass crossover filters for the drivers as well? And then would you need the FIR filter for the whole system the way you have done it? (I've read a lot and watched a ton of videos about what FIR filters are...the equations, etc, but none about the practical usage of them). I will try it on a 3-way in Sigma Studio (analog devices). Any suggestions would be great. ;)

Thanks Nathan. Fascinating, and very well explained. It would be great if you would do something about using free FIR creators, in conjunction with one of the cheaper processors, like Thomann's the.rack FIR processor.

what is the FIR software？

Nathan can you please do a full video on what we need to do all of this in the field. Processors… so on and so forth

Do you use the auto phase adjust? Is there a reason not to raise the resolution since it doesn't seem to add delay in the processor?

BTW the warning message at 3m40s relates to the largest magnitude of the FIR filter coefficients. Here the largest value coefficient/s are slightly outside the range -1.0 to 1.0 which is fine for most DSP's and when exporting the FIR filters as CSV, TXT, BIN or some of the other proprietary formats, but will be a problem for WAV file export because the WAV files are limited to values in the range -1.0 to 1.0. On the Export tab we provide an output gain setting to slightly reduce the FIR filter gain and, if necessary, bring the coefficients within -1.0 to 1.0. However this gain change may need to be anticipated elsewhere in the overall DSP chain.

can you tell about line6 bodypack transmitter with measurement mic?

I am guessing here, but probaly your software uses LPC to generate your TF inverse. Try matching or increasing the size of the import TF so that you can increase the inverse order without that red sign appearing.

So you keep interchanging linear and minimum phase while talking about the filters - it’s true that minimum phase won’t incur any additional impulse delay and shift the phase response slightly, but the linear phase filters will shift your impulse. The number of taps you put in times sample rate is overall delay. Not delay per frequency, but the delay for the linear filter to work. This can be avoided in some ways but pre-ringing becomes an issue. I would be interested to see what happens when the BSS is completely bypassed via a cable versus plugged in. I think you’ll find that the impulse in smaart is shifted a bit. Is the BSS implementing minimum phase filters? There is a change in phase response on the smaart graph