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.
Hey nathan, tks a lot for the vids. Do you know which dsps are capable of importing FIR curves from FIR creator? I think my xilica XD8080 is not able to do that, right? Tks in advance ✌🏻
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.
@@nathanlively I'm not yet, but I was so impressed with the "Gunness Focusing" using an EAW UX8800 on my old KF850s and 650s, that I'd like to try some pre-processing with FIRs on my Tannoy VQNET system.
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
For a linear-phase FIR filter, I think you mean the overall delay is "half" the filter tap length times divided by sample rate times 1000 (in ms). In FIR Designer and FIR Creator, the IR peak can be adjusted using the "delay" control, so when designing linear-phase filters, it's possible to trim more off the front of the IR by lessening the "delay" number. This, of course, starts to take the filter away from perfect linear phase but you can watch the phase error on the Export tab and adjust to your satisfaction.
To be clear, unless I’m misunderstanding you, the length of the filter is derived by filter length in taps DIVIDED BY the sample rate. i.e. for 384 taps and 48kHz, it would be 384 / 48000 = 0.008 seconds, which is 8ms
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. ;)
Hi zuke, I'm not building speakers so I'm rarely using FIR filters in my daily work. I can't really comment on how they are "customarily used". I have seen entire speakers are that design with FIR filters only. No IIR filters whatsover. But yes, I think most designs use IIR filters for the separation between bands and then FIR filters afterwards to cover a specific area. CrossLite has a users group that meets weekly where you can get training on this. Also check out the videos on the Eclipse Audio YT channel
Hey Ben, great question. Here's a selection from my interview with Michael. Michael I wouldn’t necessarily classify it as a mistake, but I do caution people not to lean too heavily on the auto-correction functions: the Auto Magnitude tab and the Auto Phase tab. Nathan But that’s the most fun. It’s the single-button solution. Michael Yeah, I completely get it because suddenly everything magically goes flat and the response becomes just the way you want it. But I guess my caution is because, as you know, drivers change their behavior with level and with temperature, and a measurement taken at one spot in a room is very different to a measurement in another spot in a room. There’s so much variability in the measurement process and in the loudspeaker. A loudspeaker is a mechanical device. It wears out. It changes its behavior over time. If you start to correct for very fine grain structure that’s in your measurement, you may be correcting perfectly for one measurement location in the room on a specific day and time, but you may make things slightly worse at other points in the room or at other levels…. www.sounddesignlive.com/myth-fir-filters-always-add-a-lot-of-delay-and-are-impractical-for-live-sound/
@@nathanlively thanks Nathan. I guess my question was not just about the auto function but also if making phase adjustments if latency becomes effected to a greater degree. Also, if NOT using the auto mag function, where a bunch of fine filters are being applied, what, if any advantage does a fir filter have over a typical parametric eq block. If you are adjusting for a specific speaker in a specific room, why would some say it is only be acceptable to use a FIR filter designed specifically for a speaker in an anechoic chamber? Why would we not adjust the filter specifically for their real world environment. I get that some of the answers are in the Eclipse response. I'm interested in trying both ways in a controlled environment and directly comparing.
![5 ideas to improve Main+Sub phase alignment with Smaart® [GSwSST35]](http://i.ytimg.com/vi/-qn5e8zQgSg/mqdefault.jpg)
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.
Eclipse Audio thanks!
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
Hi blonchay123, I haven't come to a point where I am using FIR filters in the field. Are you?
Hey nathan, tks a lot for the vids.
Do you know which dsps are capable of importing FIR curves from FIR creator? I think my xilica XD8080 is not able to do that, right?
Tks in advance ✌🏻
Hey Body, you'll want to contact Eclipse software about that.
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.
Thanks Ivan. What are you using for FIR filter creation?
@@nathanlively I'm not yet, but I was so impressed with the "Gunness Focusing" using an EAW UX8800 on my old KF850s and 650s, that I'd like to try some pre-processing with FIRs on my Tannoy VQNET system.
But if they were minimum phase filters , then they are not linear, hence not FIRs, I might miss something
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
For a linear-phase FIR filter, I think you mean the overall delay is "half" the filter tap length times divided by sample rate times 1000 (in ms). In FIR Designer and FIR Creator, the IR peak can be adjusted using the "delay" control, so when designing linear-phase filters, it's possible to trim more off the front of the IR by lessening the "delay" number. This, of course, starts to take the filter away from perfect linear phase but you can watch the phase error on the Export tab and adjust to your satisfaction.
Apologies, my 5AM mistake!!!
To be clear, unless I’m misunderstanding you, the length of the filter is derived by filter length in taps DIVIDED BY the sample rate. i.e. for 384 taps and 48kHz, it would be 384 / 48000 = 0.008 seconds, which is 8ms
@@pixelmayhem Correct. My mistake typing too fast.
Hi Parker! No linear phase filters here. Only minimum phase. That was the point of the video. If I misspoke and said "linear", I meant "minimum". :)
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. ;)
Hi zuke, I'm not building speakers so I'm rarely using FIR filters in my daily work. I can't really comment on how they are "customarily used". I have seen entire speakers are that design with FIR filters only. No IIR filters whatsover. But yes, I think most designs use IIR filters for the separation between bands and then FIR filters afterwards to cover a specific area. CrossLite has a users group that meets weekly where you can get training on this. Also check out the videos on the Eclipse Audio YT channel
Great..thanks for the quick reply. I will do that!@@nathanlively
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?
Hey Ben, great question. Here's a selection from my interview with Michael.
Michael
I wouldn’t necessarily classify it as a mistake, but I do caution people not to lean too heavily on the auto-correction functions: the Auto Magnitude tab and the Auto Phase tab.
Nathan
But that’s the most fun. It’s the single-button solution.
Michael
Yeah, I completely get it because suddenly everything magically goes flat and the response becomes just the way you want it.
But I guess my caution is because, as you know, drivers change their behavior with level and with temperature, and a measurement taken at one spot in a room is very different to a measurement in another spot in a room. There’s so much variability in the measurement process and in the loudspeaker. A loudspeaker is a mechanical device. It wears out. It changes its behavior over time. If you start to correct for very fine grain structure that’s in your measurement, you may be correcting perfectly for one measurement location in the room on a specific day and time, but you may make things slightly worse at other points in the room or at other levels….
www.sounddesignlive.com/myth-fir-filters-always-add-a-lot-of-delay-and-are-impractical-for-live-sound/
@@nathanlively thanks Nathan. I guess my question was not just about the auto function but also if making phase adjustments if latency becomes effected to a greater degree. Also, if NOT using the auto mag function, where a bunch of fine filters are being applied, what, if any advantage does a fir filter have over a typical parametric eq block. If you are adjusting for a specific speaker in a specific room, why would some say it is only be acceptable to use a FIR filter designed specifically for a speaker in an anechoic chamber? Why would we not adjust the filter specifically for their real world environment. I get that some of the answers are in the Eclipse response. I'm interested in trying both ways in a controlled environment and directly comparing.
can you tell about line6 bodypack transmitter with measurement mic?
Sure! What would you like to know about it?
Nathan Lively What microphone do you use with this Line6 device?
what is the FIR software？
FIR Creator