FIR filters

Re: FIR filters

This it not what Bennett is saying.

  1. Any transient involves a wide range of frequency components (lows and highs) that must align with each other in a specific phase relationship to sum together in a way that gives rise to the rapid upward slope when viewing the amplitude versus time (i.e. like how your DAW displays a snare hit during a mix session).
  2. If you scramble the phase relationship between the various frequency components, the energy is still there at each frequency, but the arrivals are no longer coherently combining to give the rapid rise on the amplitude vs. time display. The net result is that the energy content is the same, but the peak amplitude of the signal is reduced.
  3. Since loudspeakers, which by their nature are bandpass devices that exist in a causal (i.e. time goes forward) universe, a loudspeaker is always going to alter the phase relationships of what comes out of it.
  4. We can quantify this alteration, and pre-warp the phase of the original signal to counteract the loudspeaker so that the loudspeaker's inherent limitations serve to re-align the phase back to the original desired signal.
  5. A side effect of the pre-warping on highly organized signals like transients is that the peak amplitude of the signal is usually reduced after pre-warping.
  6. Since amplifiers are usually constrained at high frequencies not by their ability to supply energy, but rather in their maximum voltage swing (i.e. amplitude), reducing the amplitude of the signal peak by pre-warping serves to open up some breathing room on the amplifier voltage swing, and therefore allows more possible peak output from an amplifier with a given voltage swing capability.

Are we talking about the same thing? I was thinking of this link that he posted earlier in the discussion, not today's post.
http://www.fulcrum-acoustic.com/ass...ogram-loudspeaker-transient-response-2005.pdf
 
Re: FIR filters

"Everywhere" is not going to be very perfect. I would suggest that it be perfect within the designated pattern of the speaker system, with dead silence everywhere else.
 
Re: FIR filters

Isn't the opposite also possible? That the end result is less output?
In the EAW case, my understanding from the link Bennett posted earlier is that the FIR is creating an *echo* signal, that cancels reflections as they come back to the mouth of the horn. Isn't this using more bandwidth and power to get less SPL?

"In the case of EAW" ... yes in many cases they did get less SPL. For example the KF730 lost about 4 dB when focused. They pushed the drivers much harder than the IIR settings, for example there was an extreme amount of VHF boost all the way up to 20KHz :) I think there was something like 24 dB difference between the low cut on the HF driver around 1K5 and 20K! I did actually reverse engineer the settings and modified them for more power :)
 
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Re: FIR filters

Sean,


  • To the product designer the FIR offers a real possibility of cost savings. As an example, you might be able to implement a passive crossover using a minimal number of reactive components (and no resistive components). Then you can drive the loudspeaker with a single amplifier channel and use the FIR to implement a loudspeaker whose performance is nearly the same as a classic biamped design, but with a lower overall bill of materials (BOM) cost.

Yes designers are doing that but in terms of sound quality that approach has hairs on it. Acoustically the low and high drivers will not sum mathematically as we would like and you can't control the out of band issues around the crossover as well as needed.
 
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Re: FIR filters

Are we talking about the same thing? I was thinking of this link that he posted earlier in the discussion, not today's post.
http://www.fulcrum-acoustic.com/ass...ogram-loudspeaker-transient-response-2005.pdf

Jack,

Yep. Wrong discussion.

That said, horns do not exhibit discrete reflections in the way that you a probably thinking. A wave in a horn flare is constantly reflecting down the flare as it propagates along the flare length. This is an extremely complicated circumstance, because the acoustic impedance of a wave moving from throat to mouth is different than that of a wave moving from mouth to throat.
 
Re: FIR filters

"In the case of EAW" ... yes in many cases they did get less SPL. For example the KF730 lost about 4 dB when focused.

Just a spitballing troll here, but it seems as if the FIR "improvement" added latency and reduced sensitivity and/or ultimate output?

Where do I sign up?
 
Re: FIR filters

Just a spitballing troll here, but it seems as if the FIR "improvement" added latency and reduced sensitivity and/or ultimate output?

Where do I sign up?

It's unwanted noise that is lost. It's like canceling out harmonic distortion. In this case it takes signal to get rid of unwanted noise.
 
Re: FIR filters

Just a spitballing troll here, but it seems as if the FIR "improvement" added latency and reduced sensitivity and/or ultimate output?

Where do I sign up?

If you want to flatten the phase response of any system, typically you are delaying the highs to match the arrivals of the lows. Even if the lows are still arriving at the same time you will have increased the apparent latency.

In addition if you want to do some deconvolution on an impulse response that has some resonance in it, it will take time especially if you want to remove all of the resonance.

Focusing will not a change sensitivity. What EAW did was force the compression driver to be flat to about 20 KHz, if you low passed it at about 16KHz you got most of your SPL back. There was also an issue between 1Khz and 2KHz which was caused by the output of the two 7” drivers taking a dive. It took a bit of fixing. http://eaw.com/docs/1_Current_Products/KF/Spec_Sheets/KF730_SPECS_revD.pdf
 
Re: FIR filters

Thanks Phil Grahm and Peter Morris for the practical answers and info.

Reading about David Gunnes (EV, EAW, Fulcrum) and his advances.
 
Re: FIR filters

Back to the basics. I've seen the phase issues that occurs at the xover frequency between 2 drivers. So in a 3 way box, with 2 xover points, what is the audible result of this phase inversion? Is this phase inversion alone worth using FIR if you have the DSP and programing ability?

Filter Phase overlay.jpg
 
Re: FIR filters

Back to the basics. I've seen the phase issues that occurs at the xover frequency between 2 drivers. So in a 3 way box, with 2 xover points, what is the audible result of this phase inversion? Is this phase inversion alone worth using FIR if you have the DSP and programing ability?

View attachment 14167

Here’s a simple answer –

The phase of the low pass and high pass bands of a LR crossover will match each other perfectly.

The frequency and phase response of the speakers will not be perfect. Accordingly the real world amplitude and phase response of the crossover plus speakers will not look like your picture.

This is one reason to use out of band PEQ. You can get each driver to behave correctly before applying the crossover.

There are further complications when you combine a horn with a cone speaker. The horn diaphragm is pressure load and the cone speaker is mass loaded – mathematically there is roughly 90 degrees of phase difference between these two.

As a rule of thumb (IIR) crossovers with slopes of 24dB per octave or less sound fine, crossovers with slopes higher than this e.g. 48 dB octave don’t sound as good.

In my experience, all things being equal (other than phase), both IIR crossovers and FIR crossovers both sound excellent. If the speaker has a flat phase response it will not really sound noticeably better, just more real.

One thing the FIR approach to processing speakers does is give you more power to correct things, you can make the amplitude and phase response more ideal and hopefully find a better compromise.
 
Re: FIR filters

related - can I use FIR filtering to minimize interference between boxes in an array? Specifically, well-behaved, but not-perfect full range boxes that are arrayed correctly, but you can still hear interference.
 
Re: FIR filters

related - can I use FIR filtering to minimize interference between boxes in an array? Specifically, well-behaved, but not-perfect full range boxes that are arrayed correctly, but you can still hear interference.

In most situations they can’t. However … the high slopes available with FIRcrossovers may minimise the lobbing that occurs with “standard” IIR crossover.This may or may not be an advantage.
http://www.prosoundweb.com/article/a_useful_tool_creating_applying_fir_filters/P2/ - see figure 8 & 9
 
Re: FIR filters

Just a spitballing troll here, but it seems as if the FIR "improvement" added latency and reduced sensitivity and/or ultimate output?

Where do I sign up?

Back when I had KF650's and upgraded to the FIR based "focused" settings, they went from being good loud rock n roll boxes to being world class SQ. The difference was NOT subtle. Same thing with KF730.

A bit of latency and reduced sensitivity was a small price to pay. If you are really running your rig close to the edge of maximum output anyway then you should probably have more rig for the gig. And the delay doesn't matter much, I'm typically delaying tops anyway for sub alignment, more than what is incurred by FIR related DSP activities. Delay would matter more for wedges than mains - I know quite a few companies are now doing FIR on their wedges, but latency is certainly a limiting factor in the extent of its use there.
 
Re: FIR filters

Back when I had KF650's and upgraded to the FIR based "focused" settings, they went from being good loud rock n roll boxes to being world class SQ. The difference was NOT subtle. Same thing with KF730.

A bit of latency and reduced sensitivity was a small price to pay. If you are really running your rig close to the edge of maximum output anyway then you should probably have more rig for the gig. And the delay doesn't matter much, I'm typically delaying tops anyway for sub alignment, more than what is incurred by FIR related DSP activities.

Exactly. Strange that I see virtually every major touring brand implement FIR processing in some portion of their DSP presets, yet on this forum, some guys act like its a bad thing, a myth, or should not be used. My take-a-way from this discussion and all the research I've done is that FIR is clearly better when used and programed properly. I look forward to getting my new FIR processed rig. Thanks.
 
Re: FIR filters

Exactly. Strange that I see virtually every major touring brand implement FIR processing in some portion of their DSP presets, yet on this forum, some guys act like its a bad thing, a myth, or should not be used. My take-a-way from this discussion and all the research I've done is that FIR is clearly better when used and programed properly. I look forward to getting my new FIR processed rig. Thanks.

I'm FIR/IIR agnostic. The primary advantage of FIR filters is the minimal impact on phase, with the trade off being latency. The main thing is to know what you want to accomplish with the filter (to start with), i.e. what Dave Gunness learned while investigating loudspeaker anomalies and which of those could be negated with pre-processing of the audio signal and which could not. If something cannot be "fixed" there is no point in pursuing it but on the flip side if something can be improved it's a matter of which tool(s) is appropriate for the job.

You're seeing more FIR filters in use because the cost of in-the-box DSP has come down to the point that manufacturers can utilize FIR filters at no greater cost than IIR filters.
 
Re: FIR filters

The primary advantage of FIR filters is the minimal impact on phase, with the trade off being latency.

The primary advantage of FIR filters is ability to manage the filter magnitude response in a much more finely grained manner than an IIR biquad. Causal, minimum phase FIR (i.e. no taps before reference time zero) are a very powerful tool, and far preferable than futzing around with a bunch of biquads trying to achieve the same goal.

Screwing with phase is a secondary process, and optional, with FIR.