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Sound Reinforcement - Forums for Live Sound Professionals - Your Displayed Name Must Be Your Real Full Name To Post In The Live Sound Forums => LAB Lounge => Topic started by: Peter Hvedstrup on June 29, 2017, 04:21:02 AM
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Hi all
I own a small tannoy VQ60 + VS218DR system using the factory DSP with the correct preset. I really like the system and it sounds great. I now have the possibility to get a Lake LM26 DSP so i can use Tannoys newest preset from the Lake Load Librery with linear phase filters in place. The thing is that i don't really have the possibility to test if it's a improvement worth the cash since there is no other VQ in Denmark. Does anyone have any expirience with this system or just moving to new settings with liniar phase filters on ther rig?
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Hi all
I own a small tannoy VQ60 + VS218DR system using the factory DSP with the correct preset. I really like the system and it sounds great. I now have the possibility to get a Lake LM26 DSP so i can use Tannoys newest preset from the Lake Load Librery with linear phase filters in place. The thing is that i don't really have the possibility to test if it's a improvement worth the cash since there is no other VQ in Denmark. Does anyone have any expirience with this system or just moving to new settings with liniar phase filters on ther rig?
Everything is a compromise.
Linear phase filters, FIR filters etc can improve the phase response of a system.
However-they will introduce additional delay to the signal, sometimes A LOT (depending on the freq).
Is this acceptable to YOUR situation? In some case it is acceptable, in other cases not at all.
So it depends.
In many cases there is more to "the total answer" than what shows up on the computer screen.
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Hi all
I own a small tannoy VQ60 + VS218DR system using the factory DSP with the correct preset. I really like the system and it sounds great. I now have the possibility to get a Lake LM26 DSP so i can use Tannoys newest preset from the Lake Load Librery with linear phase filters in place. The thing is that i don't really have the possibility to test if it's a improvement worth the cash since there is no other VQ in Denmark. Does anyone have any expirience with this system or just moving to new settings with liniar phase filters on ther rig?
Hi Peter,
As Ivan said the FIR filters will add some extra delay but in general for FOH application this is never an issue … and your VQ60 + VS218DR system is I assume, FOH.
You can achieve a flatter amplitude and phase response than you have ever had. This will make your vocal sound more real and very clear. Each instrument and voice will be easier to separate in the mix. Your highs, mids and lows won’t really sound any better (I hope this makes sense), but things will be clearer and more natural, more defined and separate in the mix. I think it’s easier for your ears and brain to process these signals and separate the instruments when their phase response is relatively flat.
This is where most manufactures are going.
Last week I just up-grade our budget line-array boxes firmware, dB Technologies T12s and T8’s with FIR settings and they sounded and arrayed better than ever – amassing performance for what they are. The upgrade added 2ms. Today I measure the performance of our RCF TT22a-mk2’s which are now using FIR filters – very impressive.
I was also involved with the development of the FIR settings that Turbosound (before MG) use in their Flex Array, that add 2.5ms on the Lake. At the moment I trying to develop new and improved settings for these … (I hope) … the initial test are promising.
I sure everyone’s mileage will differ, but for me I thinks it’s worth it. (I'm assuming Tannoy have done a good job)
BTW Ivan, I so want to put some FIR’s on some of your Danley speakers :)
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BTW Ivan, I so want to put some FIR’s on some of your Danley speakers :)
Then do it :)
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Then do it :)
I have. For SH46HO and SH96HO.
To the OP, no. Linear phase filters have rather limited application to audio. General FIR filters have many useful properties that make them very useful for audio. Do not assume that FIR is only used to implement linear phase.
Best,
--Frank
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I have. For SH46HO and SH96HO.
To the OP, no. Linear phase filters have rather limited application to audio. General FIR filters have many useful properties that make them very useful for audio. Do not assume that FIR is only used to implement linear phase.
Best,
--Frank
And how much difference did it make?
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I have arranged to borrow the Lake LM26 and try it out before purchase.
I will do a simple A/B test to decide if it's worth it for me.
I'm used to Armonia which I find very intuitive but the Lake software just baffles me. It pushes my i7, 16gb ram and SSD pc to a halt when doing simple stuff. I don't find it intuitive at all, but maybe it's just me.
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Then do it :)
Just need some boxes. Not a lot of them in Oz, I was even going to Fly to Singapore to check out the J94s ...
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I have arranged to borrow the Lake LM26 and try it out before purchase.
I will do a simple A/B test to decide if it's worth it for me.
I'm used to Armonia which I find very intuitive but the Lake software just baffles me. It pushes my i7, 16gb ram and SSD pc to a halt when doing simple stuff. I don't find it intuitive at all, but maybe it's just me.
As much as I want to love the Lake interface as it started life hear in Adelaide Australia … and I did know and respect the guy that wrote the software, I have to agree with you about the interface.
I suggest looking at some of the YouTube videos. e.g. https://www.youtube.com/watch?v=zgK7k7iNjtc
It’s also worth remembering that the software was written to be driven on tablet out in the field.
I would be very interested in how the FIR processing goes on the Tannoys.
http://labgruppen.com/about/about-lake
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As much as I want to love the Lake interface as it started life hear in Adelaide Australia … and I did know and respect the guy that wrote the software, I have to agree with you about the interface.
I suggest looking at some of the YouTube videos. e.g. https://www.youtube.com/watch?v=zgK7k7iNjtc
It’s also worth remembering that the software was written to be driven on tablet out in the field.
I would be very interested in how the FIR processing goes on the Tannoys.
http://labgruppen.com/about/about-lake
Thanks!
Have been playing with virtual frames and think i understand the concept. I have loaded the correct preset from Load Librery and all looks good but i don't really understans why output channel gains are limited/locked to certain intervals that makes it impossible to reach the same gain structure as i have now.
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Thanks!
Have been playing with virtual frames and think i understand the concept. I have loaded the correct preset from Load Librery and all looks good but i don't really understand why output channel gains are limited/locked to certain intervals that makes it impossible to reach the same gain structure as i have now.
Normally the gains can be moved anywhere, but they can be locked or their travel limited by the speaker / setting developer to ensure consistency. Depending on how much the setting parameters have been locked it still maybe possible (unlikely) to access the gains.
Modules (highlight module) ... I/O Config ... level design
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And how much difference did it make?
That is the important question and, of course, it depends.
Does applying EQ to make the system spatial-average anechoic response (over some range of angles) approximately flat make a difference compared with not using any processing at all? Yes, it makes a big difference, and a beneficial one, in my opinion.
Does using an FIR filter, as opposed to an artfully adjusted bank of IIR filters achieving the same magnitude response, make a difference? Probably not much, when one speaker is used in isolation.
Does using an FIR filter to flatten the phase make a difference? For one speaker by itself, not much, at least on normal musical program. Theory, and the experience of more seasoned practitioners, however, says yes, when multiple speakers are required to play together.
We've been over a lot of this before on these forums. It would be nice if someone knows of a good article or two that summerize this.
Best,
--Frank
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I believe the biggest problem with fir filters are with bass cabs as the delays can be come quite significant. As with all systems it depends.
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FWIW – Almost all of my boxes are using FIR processing now – our DB Technologies DVAT12 & T8s, our RCF TT22aMK2’s, Turbososund Flex Array (just updated the settings last week to produce a better and flatter phase response - hope they work ...) and our own proprietary 8” line-array, double10’s, double 12s’ and double 14” boxes.
What’s interesting is that we always get the same comment from the audience members; they always describe the sound was very clear, and they always use the word "clear".
It was only 3 weeks ago we upgrade the DVA’s firmware; before then people liked the way we got them to sound, they might say the sound was good, nice and crisp etc. but after the upgrade everyone’s first comment was clear, the sound was very clear.
I think it has a lot to do with the phase response and how our brains process things, how we are able to focus and listen to one person even if there is extreme background noise. The flatter the amplitude and phase response the easier it is for our brains to do this. There are of course many other factors, but I believe a flat phase response over the critical vocal region does help.
FIR’s won’t make the HF sound any crisper or the mids any smoother, but it does make things clearer and easier to separate the individual voices and instruments in a mix, the difference is subtle but noticeable.
To me it’s all about processing the system in a manner that compliments the way our hearing works.
Here is a bit of text I wrote for an article, although not specifically about phase it may help explain a little:-
“… a basic understanding of psychoacoustics is helpful, especially with respect to how perceive sound in complex environments.
Our perception of sound is amazing. We can easily detect the direction of a sound source often with extreme accuracy. In an environment such as a loud party where there is an impossible amount of background noise, we can focus on one person and listen to them while ignoring all the other sounds and distractions. If we replaced our ears with a perfect microphone and played back a recording of the party we would most likely not be able to understand what that person was saying at all. The microphone cannot differentiate the direction of the sound, but our binaural hearing can.
We could also analyse the sound with an FFT measurement system. These systems are designed to measure the sound and exclude the reverberant energy from the measurements within the constraints of the sample time window length and shape (Hamming, Blackman etc.). While this would certainly tell us a lot about what’s happening the results will not match our hearing.
Our ears are mechanical devices that have resonances; in particular the basilar membrane, the base for the sensory cells of hearing within the inner ear is a pseudo-resonant structure. Its behaviour can be modelled with an auditory filter bank based on the gammatone function; put simply, other than continuous tones, what we hear is not the same as the microphone.
Our ears are able to selectively ignore many sounds, but they also take into consideration (approximately) the first 30ms of early arrivals. When there are two sounds separated by a short time delay, provided it’s below our echo threshold, we perceive a single fused auditory image. It’s a bit like a movie where provided the frame rate is greater than 16 frames per second we perceive things as a continuous motion, not a bunch of flashing pictures.
If you have ever been in an anechoic chamber and talked to someone you will instantly notice how dry and lacking in warmth the sound is. When we step outside the anechoic chamber the warmth returns to our voice as we hear the bass boost from floor, other nearby boundaries and those early reflections.”
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I believe the biggest problem with fir filters are with bass cabs as the delays can be come quite significant. As with all systems it depends.
In general all systems have the bass delayed with respect to the HF. What you can do with FIR filters and crossovers is delay the HF to match the LF delay. Typically what is done is a compromise, IIR crossovers for the subs and FIR for the rest resulting in a flat phase response from about 300Hz … like this.
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I've been playing with FIR alot, and will offer my observations....... which are constantly evolving and subject to change, so they are definitely FWIW.....
The real benefit of FIR IMO, is making phase relationships easier to coincide, between drivers or multiple speakers throughout the ranges they need to sum.
That accomplished, I completely agree with Peter's comments re clarity.
Sound gets clearer and transients get sharper....
....I figure it's because impulse response gets improved, and there is smoother magnitude through summation ranges.
The clarity gains sound a lot like the difference in compressed vs uncompressed, or like using a clearly better AD/DA conversion, etc.
And like guys have been saying in this thread, the higher the frequencies involved, the less FIR time it takes.... real gains up high are pretty easy to accomplish with minimal additional delay.
Low frequencies typically require too much time to allow much phase correction below 100-300 Hz.
I've found about 21ms (2048 taps, centered, @ 48KHz sampling) to be the minimum amount of FIR for keeping phase rotation to less than 180 degrees through sub range.
Which is just too much delay for live sound, so like Peter says, use IIR down low, and let phase roll to where phase needs to roll...
It's a damn shame though, because for playback only, I use 6144 taps (64ms delay), and get very nice flat phase through sub range, which does tighten up the sound of bass.
It kinda make the bass seem less, until you realize it's just cleaner. And then you get to crank it a bit more, with better dynamics, and an overall clearer sound.
I don't really see FIR as all that valuable for making magnitude (EQ) adjustments to an individual driver's or speaker's response.
Because what can actually be corrected is usually minimum phase (IIR), and needs to be corrected with IIR.
When IIR fixes magnitude, it also fixes phase.
As folks say, you can use FIR to make IIR adjustments.
And one benefit to FIR is you can embed a go-zillion IIR filters into a single FIR file .....
....but I'm really doubting how much fine IIR tuning actually gets done this way, unless you use a boat load of taps.
I've gotten identical looking flat magnitude and phase traces using 6144 taps and 65,000 taps per channel on a 4-way system, with tap count being the only variable.
I used maybe up to 30 embedded IIR filters on each channel, trying to minutely adjust magnitude variations. (without using excessive gain or Q)
6144 taps sounds like what I'm used to hearing with a speaker tuned solely with IIR, just considerably cleaner.
65,000 taps simply has sound details I didn't know were in the songs....kinda both amazing and unsettling at times.
I think that the smoothing on our magnitude transfers is hiding a whole lot of what we can really hear.
So all in all, at this time, when I think of using FIR, it's mostly about phase.
I give myself 10.7ms (1024 taps) to play with on mains, figuring since I'm using horn loaded subs I have that much "free FIR time" since the mains need about that much delay to begin with.
This basically allows the use of linear phase crossovers without slippage, from any freq 200Hz up at 48dB/oct. Real easy to achieve flat phase for driver summation above 200Hz.
IME, Linear phase crossovers are a true joy to use
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Is the amount of delay that the FIR filters add because of processing time? In other words if the DSP were to be faster will is have less propagation delay or have we reach the point of no return?
Also I have heard it said that a speaker system with FIR filters is more stable. For example if a pastor with a wireless mic were to be walking around out in front of the main cluster they are likely to hit spots that will feedback in a non-FIR filtered system, so it’s not consistent over the coverage area. And the same pastor walking around under the FIR filtered system any feedback points are likely to not change as much in relation to where they walk. Is this true?
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deleted owing to possible misinformation.
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never mind....
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Is the amount of delay that the FIR filters add because of processing time? In other words if the DSP were to be faster will is have less propagation delay or have we reach the point of no return?
No the delay is related to number of taps in the filter that calculates the transfer function (and sample rate). While faster processing could support crunching more numbers. I have no idea where we are wrt that being a practical limitation (my guess is not).
Also I have heard it said that a speaker system with FIR filters is more stable. For example if a pastor with a wireless mic were to be walking around out in front of the main cluster they are likely to hit spots that will feedback in a non-FIR filtered system, so it’s not consistent over the coverage area. And the same pastor walking around under the FIR filtered system any feedback points are likely to not change as much in relation to where they walk. Is this true?
Good question... If a FIR filter has a different phase response (for the same amplitude response), it could make a difference in the margin to specific feedback modes.
To support feedback the gain must be more than unity for an audio signal making multiple transits over the physical path between mic and speaker (over and over again) possessing a wavelength such that an even number of cycles reinforces itself with each pass. Phase shift will change the effective wavelength for the audio modes to reinforce and build up to run-away feedback. So it could help or it could hurt if that altered effective wavelength (due to different phase shift) is hotter than at the unaltered wavelength.
CAVEAT LECTOR, I am not the FIR filter guru here, but I do know a little about feedback.
JR
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Is the amount of delay that the FIR filters add because of processing time? In other words if the DSP were to be faster will is have less propagation delay or have we reach the point of no return?
Also I have heard it said that a speaker system with FIR filters is more stable. For example if a pastor with a wireless mic were to be walking around out in front of the main cluster they are likely to hit spots that will feedback in a non-FIR filtered system, so it’s not consistent over the coverage area. And the same pastor walking around under the FIR filtered system any feedback points are likely to not change as much in relation to where they walk. Is this true?
Hi Kevin, my understanding is there is some processing time that comes from CPU/FGPA/dsp utilization, but that is usually pretty minimal and relatively fixed.
The delay that's talked about is number of samples used by the FIR filter, often called taps...(sorry for using unexplained nomenclature.)
So for instance, the 1024 samples I use for live sound, represents a delay of 1024 samples /48,000 sampling freq or 21.3ms....but the most common use is to use impulse centering which means you end up with 1/2 the just calculated delay or 21.3/2=10.7ms.
You'd think increasing DSP speed would cut down delay, but what happens is when you double the processing speed (say 48k to 96k) you also double the nyquist freq, or rather the freq range the samples are working over.
So without doubling the number of taps used also, your FIR filter ends up with half the resolution you started with at 48kHz. Once you double the taps for the higher processing speed, you're back to the same delay.
The only thing I understand about stability, is that IIR filters with their infinite feedback loops, have the capability of doing just that...going unstable from feedback...
Maybe electro/acoustic feedback from the pastors mic increases the probability of IIR processing feedback ?.... just spit balling here...
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Also I have heard it said that a speaker system with FIR filters is more stable. For example if a pastor with a wireless mic were to be walking around out in front of the main cluster they are likely to hit spots that will feedback in a non-FIR filtered system, so it’s not consistent over the coverage area. And the same pastor walking around under the FIR filtered system any feedback points are likely to not change as much in relation to where they walk. Is this true?
That has more to do with the actual cabinets being used, how many, what sort of pattern control they have, how they are arrayed, or whether it is a single cabinet.
There are SOME things that "fancy DSP" can fix, and there are other things that they cannot fix-at least in more than 1 location at the same time.
It is one thing to fix a single cabinet on axis, but when you put that cabinet with others, getting them to work together is quite another challenge-except in a single location.
Once again, the reason a single cabinet source is the best solution. You don't have other cabinets to mess with and how they are interacting with each other.
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I think it has a lot to do with the phase response and how our brains process things, how we are able to focus and listen to one person even if there is extreme background noise. The flatter the amplitude and phase response the easier it is for our brains to do this. There are of course many other factors, but I believe a flat phase response over the critical vocal region does help.
This is all well and good. But before we get too carried away it might be instructive to do some listening tests. Real, blinded, listening tests, to reveal just how audible smoothly varying phase is.
Smoothly lagging phase is what we're talking about as the rapidly varying phase, "phase grass" as I call it, is minimum phase, and is taken out by flattening the magnitude, which can be done with a variety of causal, minimum-phase filters, FIR or IIR. What FIR filters alone allow is compensating for lagging phase, as only a non-causal filter can do. We get around the causality problem by introducing a processing delay.
I propose we tune a system to our liking and then introduce a second-order all-pass into the signal chain (the whole signal, not some pass-band that will mess with the crossovers) at a variety of frequencies and see if anyone can hear a difference. We would not be the first to try this, by the way.
To be clear, I'm not bashing the use if FIR filters. Being able to manipulate the phase in both directions (at high frequencies at least) is useful for getting smooth crossovers and getting different speakers to play together. FIR filters are easier to synthesize automatically than banks of bi-quads, even if they are just used to implement (causal) minimum-phase filters (which don't introduce any processing delay).
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Now, there are two things that continue to bother me in these discussions. First is the apparent belief that FIR filters are inherently "linear phase" and that they always introduce a processing delay. Neither is true. Filters have linear phase if they have a symmetrical impulse response, and hence are non-causal (and require a processing delay), be they FIR or IIR. (Homework problem: how do we get an IIR filter to have a symmetrical impulse response? Hint: it won't work for live sound :) )
Second is that compensating for smoothly lagging phase in an overall system response has a dramatic effect on sound quality. My belief is that you need a pretty carefully chosen test signal (try a 100 Hz triangle wave) for it to be audible. But I'm happy to be convinced otherwise.
On a more general subject, I think the real art in speaker tuning starts with observing what is relevant to various aspects of sound quality and what of that can be corrected electrically. Much as tuning the room sometimes requires a bulldozer, tuning the speaker often requires a table saw. Given that, what matters is taking the right measurements (spatial sampling rate), combining them in a certain way (averaging, throwing out outliers, weighting total isotropic power vs sound pressure at a point), and deciding what can be and needs to be corrected (smoothing, don't care regions). And as they say in medicine, "first, do no harm", or something like that.
Best,
--Frank
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This is all well and good. But before we get too carried away it might be instructive to do some listening tests. Real, blinded, listening tests, to reveal just how audible smoothly varying phase is.
Smoothly lagging phase is what we're talking about as the rapidly varying phase, "phase grass" as I call it, is minimum phase, and is taken out by flattening the magnitude, which can be done with a variety of causal, minimum-phase filters, FIR or IIR. What FIR filters alone allow is compensating for lagging phase, as only a non-causal filter can do. We get around the causality problem by introducing a processing delay.
I propose we tune a system to our liking and then introduce a second-order all-pass into the signal chain (the whole signal, not some pass-band that will mess with the crossovers) at a variety of frequencies and see if anyone can hear a difference. We would not be the first to try this, by the way.
To be clear, I'm not bashing the use if FIR filters. Being able to manipulate the phase in both directions (at high frequencies at least) is useful for getting smooth crossovers and getting different speakers to play together. FIR filters are easier to synthesize automatically than banks of bi-quads, even if they are just used to implement (causal) minimum-phase filters (which don't introduce any processing delay).
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Now, there are two things that continue to bother me in these discussions. First is the apparent belief that FIR filters are inherently "linear phase" and that they always introduce a processing delay. Neither is true. Filters have linear phase if they have a symmetrical impulse response, and hence are non-causal (and require a processing delay), be they FIR or IIR. (Homework problem: how do we get an IIR filter to have a symmetrical impulse response? Hint: it won't work for live sound :) )
Second is that compensating for smoothly lagging phase in an overall system response has a dramatic effect on sound quality. My belief is that you need a pretty carefully chosen test signal (try a 100 Hz triangle wave) for it to be audible. But I'm happy to be convinced otherwise.
On a more general subject, I think the real art in speaker tuning starts with observing what is relevant to various aspects of sound quality and what of that can be corrected electrically. Much as tuning the room sometimes requires a bulldozer, tuning the speaker often requires a table saw. Given that, what matters is taking the right measurements (spatial sampling rate), combining them in a certain way (averaging, throwing out outliers, weighting total isotropic power vs sound pressure at a point), and deciding what can be and needs to be corrected (smoothing, don't care regions). And as they say in medicine, "first, do no harm", or something like that.
Best,
--Frank
Hi Frank,
Can you hear it … I have done some tests using the same speaker with “normal” IIR 24dB LR filters and then with matching linear phase FIR crossovers + some all-pass filters to fix a few small bumps making it flat from 300-400Hz up.
The conclusion I came to as I said above “Your highs, mids and lows won’t really sound any better (I hope this makes sense), but things will be clearer and more natural, more defined and separate in the mix.” What they do is subtle, but noticeable.
I think part of the problem regarding tests to determine if we can hear phase distortion is has been using continuous tones. Our hearing is extremely sensitive to first arrival times yet all this information is processed through a pseudo-resonant structure whose behaviour can be modelled with an auditory filter bank based on the gammatone function (as I noted above).
Without going into detail, I believe this means we can detect a lot more about the nature of random transient sounds than a continuous tone; after all this is what we were designed to do and one of the mechanisms that kept us safe from predators.
In our case (live sound) the critical thing we need to do is hear the human voice or voices within the mix as clear as possible …
http://www.audioholics.com/room-acoustics/human-hearing-phase-distortion-audibility-part-2
In this respect, dare I say, Dr Floyd Toole got it wrong to some extent … I’m with Lipshitz who said “Using pre-recorded music (male & female singing) fed through a 2nd-order all-pass network, with a Q of .5, audible effects were noted with a 95% confidence level. Using a variety of unpitched sounds recorded anechoically, phase effects were again audible.”
The other thing to note is most of these tests were done with 2nd order networks, while we typically use 4th order filters in our speakers, I believe this make things much more noticeable. I don’t think a smooth slowly varying phase is that noticeable.
A speaker designer can also do a lot more tricks with FIR processing than just flatting the phase. You can use extremely high crossover slopes to minimise radiation errors or limit effect of out band problems that your transducers may have, and you have much greater ability to finely correct amplitude response issues.
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Is the amount of delay that the FIR filters add because of processing time? In other words if the DSP were to be faster will is have less propagation delay or have we reach the point of no return?
Also I have heard it said that a speaker system with FIR filters is more stable. For example if a pastor with a wireless mic were to be walking around out in front of the main cluster they are likely to hit spots that will feedback in a non-FIR filtered system, so it’s not consistent over the coverage area. And the same pastor walking around under the FIR filtered system any feedback points are likely to not change as much in relation to where they walk. Is this true?
If you what to use an FIR filter to implement a linear phase crossover it will take time. The time taken is proportional to the crossover frequency; the lower it is the longer you need. If you had a perfect DSP with zero processing latency it would still take time because of the way the filters are implemented.
If you use an IIR or analogue crossover, the low frequencies are delayed. You can use an FIR filter to flatten the phase so that the HF delay matches the sub delay … and like the FIR crossover the time you end up delaying things is proportional to the crossover frequency, the lower the crossover frequency the more you have to delay the HF to match the initial delay caused by the crossover.
As a generalization, the speakers I have used with FIR filters do behave better in terms of feedback. I think this is due in part to being able use the FIR filters to correct for amplitude and time domain issues better than ever before. They can also be used to minimize radiation errors around the crossover frequencies.
http://dolby.invisionzone.com/index.php?app=core&module=attach§ion=attach&attach_id=8 see pages 8 to 12
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Hi Peter,
Responses inserted below. (Hope all the quotes work.)
Can you hear it …
In my limited (unscientific) experiments, for recorded music, no, not so far. Interestingly, on certain continuous test signals, such as the 100 Hz triangle wave I alluded to, yes. But just because I can't hear it doesn't mean that someone else can't. And I might hear it under different circumstances. There are many things to confound such tests, such as, I'm guessing, a small-room reverberant environment with ample early reflections.
Without going into detail, I believe this means we can detect a lot more about the nature of random transient sounds than a continuous tone; after all this is what we were designed to do and one of the mechanisms that kept us safe from predators.
Indeed. I remember hearing a talk in which research came up suggesting that there is an inverse relationship between distinct pitch perception (critical bandwidth) and time discrimination in various species. As I recall domestic cats have only about 5 critical bands but have good enough localization (time discrimination) to be able to pounce on a mouse in total darkness guided only by its sounds.
In our case (live sound) the critical thing we need to do is hear the human voice or voices within the mix as clear as possible …
Yes, agree. Get speech intelligibility right and most of the rest will follow. OK, not the boom-boom from the subs :)
The other thing to note is most of these tests were done with 2nd order networks, while we typically use 4th order filters in our speakers, I believe this make things much more noticeable. I don’t think a smooth slowly varying phase is that noticeable.
Yes, and speakers themselves, even if you ignore all the wiggles, are >> 4th order networks. So there is much to mess with.
A speaker designer can also do a lot more tricks with FIR processing than just flatting the phase. You can use extremely high crossover slopes to minimize radiation errors or limit effect of out band problems that your transducers may have, and you have much greater ability to finely correct amplitude response issues.
I've experimented with high-slope crossovers a little. There no doubt are situations where they work well but for the time being I'm getting along pretty well with 4th or lower order crossover filters as I'm working mostly with Danley boxes or coaxes where radiation errors (polar weirdness around the crossover) are less of a problem. (Not counting the order of any sub-band FIR filters which have an order equal to the number of taps - 1.)
This all started when I noted that I'd worked on FIR tunings for some Danley products, and their philosophy seems very much to be low slope, wide overlap when it comes to crossovers. And they know a lot more about their speakers than I know about any speakers.
Best,
--Frank
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Hi guys,
Can you hear it?
My take, where i feel pretty secure.....
On well recorded music......When I use enough FIR to flatten phase all the way through the sub to main summation region (down to about 60Hz)...YES, I can hear it, and everyone I've asked to listen critically says they can too.
It's simply clearer with a very real wow factor mono. And stereo imaging gets much, much more solid. Wow again.
Most of my critical listening/measuring has been using the PM60s, and by enough FIR, I mean 64ms impulse centered, because that's what my minidsp units deliver.
I can tell by sims, and then measure as well, that even with these 6144 taps, there is still a little slippage mag and phase, vs hoped for correction, once below 100Hz.
I do not know if flat phase is really better than mildly sloping phase (ala all pass A/B comparisons, or for another comparison, vs meyer signature phase traces)...I kinda doubt it.
I do know phase matching throughout x-over summation matters, if nothing more than for magnitude summation....heck, we all know that.
IMO, linear flat phase just makes it easy and guarantees summation, even when we move x-over freq up or down.
I also know, that in trying to learn how to pare down delay, or chop taps, for live sound...the wow factor starts disappearing...dishearteningly so.
I was bumming last weekend at the reduction in SQ, as I lowered tap count/ changed processing, on the PM90s to get set up for a live show..so I gotta learn more here....
Which begs......Peter, I would love to know the tricks you've learned at phase alignment, while keeping delay minimized.
I get IIR for x-over down low. But I don't see how you get get any kind of brickwall filtering at any freq below up way high (say 6300 -:).
And I need to get a better grip on how to use all pass filters....
It amazes me what you get done with a couple of ms......pls help ! :)
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Hi guys,
Can you hear it?
My take, where i feel pretty secure.....
On well recorded music......When I use enough FIR to flatten phase all the way through the sub to main summation region (down to about 60Hz)...YES, I can hear it, and everyone I've asked to listen critically says they can too.
It's simply clearer with a very real wow factor mono. And stereo imaging gets much, much more solid. Wow again.
Most of my critical listening/measuring has been using the PM60s, and by enough FIR, I mean 64ms impulse centered, because that's what my minidsp units deliver.
I can tell by sims, and then measure as well, that even with these 6144 taps, there is still a little slippage mag and phase, vs hoped for correction, once below 100Hz.
I do not know if flat phase is really better than mildly sloping phase (ala all pass A/B comparisons, or for another comparison, vs meyer signature phase traces)...I kinda doubt it.
I do know phase matching throughout x-over summation matters, if nothing more than for magnitude summation....heck, we all know that.
IMO, linear flat phase just makes it easy and guarantees summation, even when we move x-over freq up or down.
I also know, that in trying to learn how to pare down delay, or chop taps, for live sound...the wow factor starts disappearing...dishearteningly so.
I was bumming last weekend at the reduction in SQ, as I lowered tap count/ changed processing, on the PM90s to get set up for a live show..so I gotta learn more here....
Which begs......Peter, I would love to know the tricks you've learned at phase alignment, while keeping delay minimized.
I get IIR for x-over down low. But I don't see how you get get any kind of brickwall filtering at any freq below up way high (say 6300 -:).
And I need to get a better grip on how to use all pass filters....
It amazes me what you get done with a couple of ms......pls help ! :)
Hi Mark,
With the Lake set to a FIR processing time of 2.5ms you can have a 24 or 48 dB/oct LR shaped linear phase crossover down to 500Hz. At 6K you can have a brick wall – ish 93 dB/oct. At 3.5ms I can achieve 80dB/oct at 1.5KHz.
The Lake does not let you manipulate the phase separately nor enter your own tap coefficients. To flatten the phase you need to use all-pass filters … it’s tricky but you can make it work.
If you can ensure your drivers are behaving well out side of their operating band there is often no need or advantage using brick wall filters. On the boxes you mentioned, the PM60 & 90 I used 24dB linear phase crossover filters on the mid-range but as steep as I could get on the VHF because of the way the VHF driver has been designed and behaves.
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It's a damn shame though, because for playback only, I use 6144 taps (64ms delay), and get very nice flat phase through sub range, which does tighten up the sound of bass.
It kinda make the bass seem less, until you realize it's just cleaner. And then you get to crank it a bit more, with better dynamics, and an overall clearer sound.
This is something my students and I conclude unanimous in class time and again.
According to Coda Audio, it's also a reoccurring "complaint" with respect to their sensor controlled subwoofers featuring less phase shift due to lack of an electronic HPF for speaker protection.
I'm starting to suspect that the audibility of phase distortion is somehow tied to harmonic distortion. Distorted sound is typically perceived as being louder over clean sound.
A lot of research regarding phase distortion has been done using square waves or transient signals in general. Phase distortion regarding the former, manifests itself as spectral or timbral change, altering the sequence of the harmonics (at least in my mind).
In general there's good consensus that less phase distortion sounds clearer. All loudspeakers ultimately distort (often excursion related), audibly or inaudible. Regardless, this is evidently the moment where sine waves become square waves. Systems with more phase distortion (relatively speaking) alter the sequence of harmonics which is perceived as spectral or timbral change making them sound less clear.
I surmise auditory masking (including temporal) plays an important part here, as well as the perception of loudness over time (integration).
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This is something my students and I conclude unanimous in class time and again.
According to Coda Audio, it's also a reoccurring "complaint" with respect to their sensor controlled subwoofers featuring less phase shift due to lack of an electronic HPF for speaker protection.
I'm starting to suspect that the audibility of phase distortion is somehow tied to harmonic distortion. Distorted sound is typically perceived as being louder over clean sound.
A lot of research regarding phase distortion has been done using square waves or transient signals in general. Phase distortion regarding the former, manifests itself as spectral or timbral change, altering the sequence of the harmonics (at least in my mind).
In general there's good consensus that less phase distortion sounds clearer. All loudspeakers ultimately distort (often excursion related), audibly or inaudible. Regardless, this is evidently the moment where sine waves become square waves. Systems with more phase distortion (relatively speaking) alter the sequence of harmonics which is perceived as spectral or timbral change making them sound less clear.
I surmise auditory masking (including temporal) plays an important part here, as well as the perception of loudness over time (integration).
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Hi Merlijn,
I have had a similar experience with one of my designs where I have pushed the flat phase response to what I consider a practical limit. Linear phase crossovers on all pass bands and no HPF on the sub (you just have to be careful selecting your kick drum mic so it does not drive the subs too low). The sub is a ported design but this and its natural roll off are the only things that impact the phase response. I have managed to contain the total latency to about 15ms so the system is still usable for FOH applications. Despite the sub being a double 21” people are amazed at how tight and punchy the low frequencies are.
(must check out Coda's sub)
I total agree about changing the sequence of harmonics. If we consider several people signing together; if we can keep the timing of the harmonics that make out each voice linear then I believe our hearing / brain is able to determine much more easily which harmonics are associated with which fundamental – this results in the sound becoming clearer and each voice is separated in the mix … and the music becomes more enjoyable because we don’t have to concentrate so hard.
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I'm sure you guys know this but loudspeaker harmonic distortion in the LF region that expresses octaves higher than the fundamentals will be much more easily heard (as taught by Fletcher-Munson constant loudness curves). Some customers are so accustomed to bottom octave distortion that they need to re-learn what (clean) linear reproduction sounds like.
[speculation]
I try not to argue with other people about what they hear but I expect LF phase response to be low on the list for audibility (all else equal), but if that phase response changes interaction with room boundaries or standing waves, that could cause very audible amplitude changes.
[/speculation]
Or not... of course better is always better, as long as it doesn't make something else worse. I am not the speaker expert here (obviously).
JR
PS: Music playback and loudspeaker design is a bundle of compromises. Experienced listeners may favor some metrics over others due to training.
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If you can ensure your drivers are behaving well out side of their operating band there is often no need or advantage using brick wall filters. On the boxes you mentioned, the PM60 & 90 I used 24dB linear phase crossover filters on the mid-range but as steep as I could get on the VHF because of the way the VHF driver has been designed and behaves.
Hi Peter, I've been using out-of-band filters, but really only to assure response in the critical x-over region, a region I've narrowed to near nothing with steep x-overs.
Between the steep filters and flat phase, I've been taking the easy way out......at the expense of delay.
I've known this all along, but now that I'm doing more live, it's time to get serious assessing the trade-offs.
Your post makes me realize i need to do all I can with out-of-band IIR, to get phase to behave as best it can through the critical region, before applying the linear phase x-overs.
Thx :)
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I'm sure you guys know this but loudspeaker harmonic distortion in the LF region that expresses octaves higher than the fundamentals will be much more easily heard (as taught by Fletcher-Munson constant loudness curves). Some customers are so accustomed to bottom octave distortion that they need to re-learn what (clean) linear reproduction sounds like.
+1
But at the same time there's auditory masking which allows us to get away with our increased sensitivity for higher bass frequencies up to a certain degree. The higher harmonics are masked by the fundamental.
IMO this explains the stringent and yet "lenient" tolerances in the CEA-2010 standard (attention potential topic swerve) which is an excellent way of rating subwoofer output.
However in typical systems, the fundamental is generally also late to the party (for subwoofers in order of 20 to 30 ms) which might result in less auditory masking due to reduced overlap compared to a system with no phase distortion.
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This is something my students and I conclude unanimous in class time and again.
According to Coda Audio, it's also a reoccurring "complaint" with respect to their sensor controlled subwoofers featuring less phase shift due to lack of an electronic HPF for speaker protection.
Hi Merlijn, I sure was glad to see this post. Sometimes I get the feeling folks think I'm nuts for all the phase talk I throw out...
They're no doubt right about the nuts part ! but at least maybe for other reasons ! :)
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I'm sure you guys know this but loudspeaker harmonic distortion in the LF region that expresses octaves higher than the fundamentals will be much more easily heard (as taught by Fletcher-Munson constant loudness curves). Some customers are so accustomed to bottom octave distortion that they need to re-learn what (clean) linear reproduction sounds like.
[speculation]
I try not to argue with other people about what they hear but I expect LF phase response to be low on the list for audibility (all else equal), but if that phase response changes interaction with room boundaries or standing waves, that could cause very audible amplitude changes.
[/speculation]
Or not... of course better is always better, as long as it doesn't make something else worse. I am not the speaker expert here (obviously).
JR
PS: Music playback and loudspeaker design is a bundle of compromises. Experienced listeners may favor some metrics over others due to training.
Hi John, I may be misunderstanding what Merlijn and Peter are saying, but I think the "harmonic distortion" being discussed is not distortion being produced by the subwoofer as we typically think about it. (Like when the sub is being driven past linearity.)
I take it to be the inability of the speaker system, the subwoofer summing with mains, to accurately reproduce the relationship between a wave and it's harmonics in a complex waveform. Please correct me if I wrong guys...
For instance, we know a square wave is a prescribed summation of a sine wave and it's odd harmonics.
Well, that summation has an implicit phase relationship between all the frequencies being summed.
If the DUT recreating the square wave isn't phase neutral across all frequencies in play, how can it accurately sum the square wave together.
Seems to me, any complex waveform has the same property.....it relies on a given phase relationship between all the waveform components.
And that given phase relationship relies on phase neutral recreation......
So another for instance, if a sub HPF is rotating phase, doesn't that mean the phase relationship between a fundamental, and lessor or non-rotated harmonic frequencies higher up, has to change?
I think this is where we might get a perceived timbrel shift, and I really think it's where we get smeared transients.
PS...Couldn't agree more re comment about experienced listeners....I know I hear more every time I toy with the How to Listen app folks often post...
http://harmanhowtolisten.blogspot.com/
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Hi John, I may be misunderstanding what Merlijn and Peter are saying, but I think the "harmonic distortion" being discussed is not distortion being produced by the subwoofer as we typically think about it. (Like when the sub is being driven past linearity.)
That was the distortion I was thinking of.
I take it to be the inability of the speaker system, the subwoofer summing with mains, to accurately reproduce the relationship between a wave and it's harmonics in a complex waveform. Please correct me if I wrong guys...
I am not sure how audible the phase relationship is? I would expect it to be secondary to the amplitude of the different partials.
For instance, we know a square wave is a prescribed summation of a sine wave and it's odd harmonics.
Well, that summation has an implicit phase relationship between all the frequencies being summed.
If the DUT recreating the square wave isn't phase neutral across all frequencies in play, how can it accurately sum the square wave together.
yes the phase relationship is important for it to "look" like a square wave, but to sound*** like one not so much (IMO). Not to mention that square waves don't exist in nature.
Seems to me, any complex waveform has the same property.....it relies on a given phase relationship between all the waveform components.
And that given phase relationship relies on phase neutral recreation......
So another for instance, if a sub HPF is rotating phase, doesn't that mean the phase relationship between a fundamental, and lessor or non-rotated harmonic frequencies higher up, has to change?
I think this is where we might get a perceived timbrel shift, and I really think it's where we get smeared transients.
If there was a hierarchy for audibility of characteristics phase shift would be low on the list. It matters a bunch in crossovers at the crossover point because both passbands overlap and phase will directly impact the summed amplitude of the same frequency coming from two drivers.
PS...Couldn't agree more re comment about experienced listeners....I know I hear more every time I toy with the How to Listen app folks often post...
http://harmanhowtolisten.blogspot.com/
I learned a long time ago that A) I can measure things I can't hear, and B) not to argue with people on the internet about what they hear. 8)
JR
*** there may be some subtle issues related to phase relationship and superposition of the partials that could cause momentary amplitude peaks to exceed the channel dynamic range, causing IMD. IIRC some clever broadcast limiters used frequency specific phase shift to reduce peaks without otherwise altering the amplitude weighting of the partials.
PS: sorry to veer off even further.
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Also I have heard it said that a speaker system with FIR filters is more stable. For example if a pastor with a wireless mic were to be walking around out in front of the main cluster they are likely to hit spots that will feedback in a non-FIR filtered system, so it’s not consistent over the coverage area. And the same pastor walking around under the FIR filtered system any feedback points are likely to not change as much in relation to where they walk. Is this true?
I've experienced this first hand, upgraded a KF750 install with UX8800 processors from MX8750. On the first show I was able to get 10-12dB more level from the talking heads on stage with omni headbands without feedback or ringing than I did the previous night on the same system with the older processors.
AFAIK FIR is included in the greybox settings on the UX8800, in this case it made a huge difference.
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I have not yet buried my head in to all this FIR stuff, but i see it more and more used in line array systems. Now i know it can make one box shine nad be perfectly flat and all that. But what happens when you array 12 of them?
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That was the distortion I was thinking of.I am not sure how audible the phase relationship is? I would expect it to be secondary to the amplitude of the different partials.yes the phase relationship is important for it to "look" like a square wave, but to sound*** like one not so much (IMO). Not to mention that square waves don't exist in nature.If there was a hierarchy for audibility of characteristics phase shift would be low on the list. It matters a bunch in crossovers at the crossover point because both passbands overlap and phase will directly impact the summed amplitude of the same frequency coming from two drivers. I learned a long time ago that A) I can measure things I can't hear, and B) not to argue with people on the internet about what they hear. 8)
JR
*** there may be some subtle issues related to phase relationship and superposition of the partials that could cause momentary amplitude peaks to exceed the channel dynamic range, causing IMD. IIRC some clever broadcast limiters used frequency specific phase shift to reduce peaks without otherwise altering the amplitude weighting of the partials.
PS: sorry to veer off even further.
Hi John,
While I would argue there is defiantly and advantage to having a flat phase response and you can hear the difference, it is very subtle compared to having a correct amplitude response and low distortion etc.
I would even argue unless you get everything else to a reasonably high standard you will not be able tell the difference between a flat or wrapped phase response.
The interesting thing that seems to happen is that once you flattened the phase, the amplitude errors and your EQ corrections become much more noticeable.
Even though square waves don’t exist in nature they do tell you a lot about the speaker – they tell you your speaker has a good amplitude response and that fundamental, and the third , fifth , seventh harmonics etc. (from the Fouries series that make up a square wave) are all arriving at the correct time.
Being able to reproduce a square wave is just a by-product of a good amplitude response and flat phase response. If your speaker can do this they should be able to match the true shape of most complex musical waveforms. I believe the relative timing of the fundamental and harmonics are important particularly on transient signals given way we and our ears process sound and results in the improved clarity we have perceived.
How we hear is a huge complex topic just by itself – this presentation about Pitch, Timbre and Source Separation maybe of interest http://player.slideplayer.com/1/679971/# (have a look around slide 30)
To me it’s hard to describe the difference when I A/B a linear and wrapped phase response other than to say the linear setting (within limits) sounds clearer and more real; this is certainly the reaction we get from the audience.
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(http://i1.kym-cdn.com/photos/images/newsfeed/000/353/279/e31.jpg)
Thanks for all your inputs and answers to my original question. I would be lying if I pretended to understand everything that was written (this is the lab lounge ;D ), but even so I am even more keen to try out the LM26 with Tannoys own preset from the Load library.
I got a hold of the LM26 last night and struggled to do even simple stuff like renaming a frame and loading the Tannoy preset. The Lake software is nothing less than horrible! In the end I think I managed to get the preset loaded and verified. To night I will do some A/B testing against my Linea Research DSP with factory IIR preset to see if the difference justifies the investment. Maybe it will be the FIR or maybe it's just a better preset or maybe I won't be able to hear any difference or it might even sound worse.
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(http://i1.kym-cdn.com/photos/images/newsfeed/000/353/279/e31.jpg)
Thanks for all your inputs and answers to my original question. I would be lying if I pretended to understand everything that was written (this is the lab lounge ;D ), but even so I am even more keen to try out the LM26 with Tannoys own preset from the Load library.
I got a hold of the LM26 last night and struggled to do even simple stuff like renaming a frame and loading the Tannoy preset. The Lake software is nothing less than horrible! In the end I think I managed to get the preset loaded and verified. To night I will do some A/B testing against my Linea Research DSP with factory IIR preset to see if the difference justifies the investment. Maybe it will be the FIR or maybe it's just a better preset or maybe I won't be able to hear any difference or it might even sound worse.
The Lake stuff is a bit tricky to use, not sure if UI was being driven by Clair Brothers when it all started or the Lake guys ... anyway love to know how you get on .... and thanks for starting a very interesting thread ;D
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To night I will do some A/B testing against my Linea Research DSP with factory IIR preset to see if the difference justifies the investment. Maybe it will be the FIR or maybe it's just a better preset or maybe I won't be able to hear any difference or it might even sound worse.
Hi Peter, the ASC48 ? Have you tried their LIR stuff?
Linea makes some pretty bold claims about it..http://linea-research.co.uk/wp-content/uploads/LR%20Download%20Assets/Tech%20Docs/LIR_LinearPhaseCrossovers.pdf
Please post any impressions if you can...
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Hi Peter, the ASC48 ? Have you tried their LIR stuff?
Linea makes some pretty bold claims about it..http://linea-research.co.uk/wp-content/uploads/LR%20Download%20Assets/Tech%20Docs/LIR_LinearPhaseCrossovers.pdf
Please post any impressions if you can...
Craig Leerman wrote an great review on the amp with this processor embedded in the current issue of LSI.
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Craig Leerman wrote an great review on the amp with this processor embedded in the current issue of LSI.
Thanks Scott
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I did the comparison between the Lake LM26 with Load Librery preset and Tannoy VNET SC1 (Linea Research lsc 2x6) with factory preset. Same amps and same gain.
The LM26 sounded better by quite a bit. I can only describe it as more defined vocals and less strain on your ears when the volume got LOUD. The VNET SC1 was more brittle and less precise. Funny because i always loved the sound of my old setup but the LM26 just makes it even better. Test tracks where Steely Dan - Jack of Speed and Massive Attack - Teardrop.
I did have problems gain matching the DSP's and the LM26 ended up with sub +5, mid 0 and hi -9. These gain settings on the VNET SC1 lacked a huge amount of sub and i had to gain it 13+ db to get any sub at all. In the preset for the VNET SC1 there is no factory EQ on sub at all but there must be in the Load Librery one to make such a huge difference.
I don't know if its up to the LM26 being superior sounding, FIR or just a better preset but i need to buy one NOW!
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Hi Peter, the ASC48 ? Have you tried their LIR stuff?
Linea makes some pretty bold claims about it..http://linea-research.co.uk/wp-content/uploads/LR%20Download%20Assets/Tech%20Docs/LIR_LinearPhaseCrossovers.pdf
Please post any impressions if you can...
No I have not tried it, its not available in my part of the world but it certainly has come to my attention.
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I don't know if its up to the LM26 being superior sounding, FIR or just a better preset but i need to buy one NOW!
The LM26 doesn't sound any better IMO. You could put the VNET SC1 IIR settings in the Lake and it would sound the same. Its the extra stuff you can do with FIR filters that make the preset better - flatter amplitude and phase response ...etc.
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I found this little video about flattening the phase, it may be of interest ... https://www.youtube.com/watch?v=oDG7lgQybzk
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An acquaintance mentioned that in trying linear phase xover the sound became less analog, more metallic, dry and analytical.
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I have the http://www.deqx.com/product-hdp4-overview.php unit at home which does this:
"DEQX processors correct the distortion that all speakers make—electro-mechanical devices that they are—and offer room compensation as an added extra. While righting frequency-response errors as other units do, they also uniquely fix critical timing errors by adjusting thousands of frequency groups so that they arrive on time.
Other ‘room correction devices’ simply adjust amplitude response (uneven output at different frequencies) to offset room modes. But they can’t address the fine details of phase coherence; the group-delay timing issues that manifest in all real-world speakers."
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The process is:
1. Measure the speaker in a condition as close as possible to an anechoic chamber (supplied Dayton EMM6 connected to the DEQX unit). DEQX recommends measuring the speaker in an open space (garden, etc). I've never done this because of the hassle involved.
2. The DEQX unit will automatically adjust the phase response and timing at each frequency tap point such that sound arrives in phase and in time at the listener (I hope I'm using the correct terminology).
3. The last step is to measure the speaker in its final position in the room and the DEQX will correct for room issues.
In our least enjoyable home system (many years ago got suckered into buying a B&O beolab 8000 by a sweet talking salesman), the DEQX truly transformed the way it sounded. I could sit and listen to a full track of music through the beolabs. I couldn't before.
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So I tried it with a pair of Danley SH50 & single TH118 (QSC PL340 for the top and QSC PL380 for the sub).
I conducted the speaker measurement indoors in a 65ft x 65ft room. Doing so in a room with reflections is not recommended and probably explained the results I got.
I let the DEQX automatically adjust the group delay and phase response automatically, and the SH50 sounded thin and dry.
I disabled the speaker adjustment and compared only LR24 vs linear-phase xover (flip of a virtual switch in the deqx) at 80 or 100hz, and the SH50 sounded thin and dry. This was unexpected.
I will try again some day but I believe it was because I did not conduct step 1 correctly. And I am not yet familiar with the minutiae of manual adjustment available in the DEQX beyond the automatic setting (which worked for my beolabs).
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If the Danleys in my project are greenlit, I will run the SH50 / DBH218 through the DEQX and see what it can do. The only caveat is the potential delay it introduces as this is a home unit for music playback.