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Does Phase Distortion/Shift Matter in Audio? (no*)

312elements

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Thank you all for your generous explanations. It seems as though focusing my question would yield better results (hopefully).

In what circumstances should one opt for FIR vs IIR? The benefit of FIR appears to be related to phase but I’ve yet to really wrap my head around use cases. On the surface, it seems like those that are a little OCD do everything with FIR because they can and the rest make do with IIR. This particular question is important to me as I try to decide between Camilla DSP and MiniDSP. I guess I’m attempting to select the right tool for the job. Both seem to have their own set of pros and cons. The learning curve for Camilla DSP seems higher but that’s not a deal breaker to me if it leads to an audibly superior sound. So in the context of DSP and digital crossovers, does phase distortion matter? I also recognize Minidsp has limited FIR capabilities.
 

gnarly

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Thank you all for your generous explanations. It seems as though focusing my question would yield better results (hopefully).

In what circumstances should one opt for FIR vs IIR? The benefit of FIR appears to be related to phase but I’ve yet to really wrap my head around use cases. On the surface, it seems like those that are a little OCD do everything with FIR because they can and the rest make do with IIR. This particular question is important to me as I try to decide between Camilla DSP and MiniDSP. I guess I’m attempting to select the right tool for the job. Both seem to have their own set of pros and cons. The learning curve for Camilla DSP seems higher but that’s not a deal breaker to me if it leads to an audibly superior sound. So in the context of DSP and digital crossovers, does phase distortion matter? I also recognize Minidsp has limited FIR capabilities.
Hi, great questions imo...the ones that matter most.
I think how deep, if, you get into FIR depends on what you want to do with it.

For multi-way DIY speaker building, I find FIR with linear-phase xovers a godsend,.... easy ....that give great on and off axis response....and come with flat phase whether it can be heard or not.
But these are 4-way and up. If I was only working on a relatively simple 2-way and sub, not so sure how much FIR would help. Just comes down to SPL and bandwidth requirements per driviver section.

For room tuning, I must say I'm not really qualified to have a strong opinion, other than say I think i understand the science better than at least 95% of folks...and have my doubts that FIR brings a lot to the party.
That said, I'm always learning and would truly love to have my doubts removed.
 

levimax

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Thank you all for your generous explanations. It seems as though focusing my question would yield better results (hopefully).

In what circumstances should one opt for FIR vs IIR? The benefit of FIR appears to be related to phase but I’ve yet to really wrap my head around use cases. On the surface, it seems like those that are a little OCD do everything with FIR because they can and the rest make do with IIR. This particular question is important to me as I try to decide between Camilla DSP and MiniDSP. I guess I’m attempting to select the right tool for the job. Both seem to have their own set of pros and cons. The learning curve for Camilla DSP seems higher but that’s not a deal breaker to me if it leads to an audibly superior sound. So in the context of DSP and digital crossovers, does phase distortion matter? I also recognize Minidsp has limited FIR capabilities.
I am in the middle of this and the answers are long and there is more than one approach. I am very much in agreement with @KSTR approach but not everyone is. Regarding Mini DSP vs Camilla I would say Mini DSP is much easier to learn and use but the computer power is very limited compared to a PC so use of FIR filters is going to be limited especially at low frequency. I found Camilla DSP very hard to learn and use even though it doesn't have to be. I would look at 2 other alternatives if using a PC either EKIO or Dephonica. Dephonica is free and EKIO is about $100. REW and Rephase are also free and invaluable.

I am using IIR filters for the crossovers (acoustic LR 4's) and then a Fir filter to "unwrap" the phase and do room EQ.

Some people are very adamant that "Linear Phase" crossovers are the only way to go.

Good luck on your learning adventure.
 

j_j

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- Minimum phase (in context of loudspeakers and rooms) - this is the only version of the measurement which is correctible by inversion, which corrects both magnitude and phase at the same time. Loudspeakers are generally considered min phase systems, although there may be regions within a loudspeaker which are non-min phase.
Actually, a linear, non-minimum phase system CAN be corrected with LONG filters and a lot of work, but the correction is never perfect. It can, however, be very, very CLOSE to perfect.

A pure delay system (i.e. linear phase) has no phase issues to be corrected.

- Minimum phase (in context of DSP) - this is the minimum amount of delay encountered by a signal when it passes through a filter.

No, it means that all poles and zeros of any filters involved are inside the unit circle. (or in the case of zeros, on it, but that is a singular system that is beyond this discussion presently)

- Linear phase (in context of DSP) - every frequency encounters the same delay when passing through a filter. Suppose this is 1ms (1/1000s). This is 10 full sine waves for a 10kHz signal, 1 sine wave for a 1kHz signal, and 1/10 of a sine wave for a 100Hz signal - i.e. there is frequency dependent phase rotation.

Linear phase means phase = 2* pi * f * t in terms of radians, hz, and time in seconds. So there is lots of phase shift, but the phase shift is a STRAIGHT LINE corresponding to the equation just referenced.

- Excess phase (in context of loudspeakers and rooms) - non-minimum phase behaviour introduced by the room, e.g. furnishings, room shape, boundaries, etc. The measured phase as captured by the microphone contains the loudspeaker's min phase response + excess phase. The EP is normally removed and discarded, as it can not be corrected (strictly speaking, not true - it can be partially corrected).

Excesss phase? You mean "non-minimum phase" really. No, the non-minimum phase part of a signal is not removed before calculating frequency response adjustment, it can't be, it's part of the problem. Even if you can't correct the phase (and you actually can with long FIR filters, to any tolerance required, and work) you DO want to correct the magnitude response that is created by it.

- Mixed phase (in context of DSP) - a DSP product that contains both IIR and FIR filters, i.e. both lin phase and min phase filters.

No, this is misuse. Any impulse response can be divided into its minimum phase, delay, and non-minimum phase parts. IIR filters have minimum phase poles, by definition, they can have non-minimum phase zeros if they want to. FIR filters can be literally anything you want, anything at all, that's within the filter length limits. MOST "FIR" filters are actually constant-delay filters, but sometimes that's actually not the right way. Remember a constant delay filter is ONLY zeros, and contains an exact duplication of every zero inside side the unit circle with a zero at 1/MP_zero. Yes. That's why it's constant delay, exactly (removing any zeros on the unit circle from this discussion, although obviously not from the filter!) mirrored in inside/outside pairs (quads usually, since most are complex roots). There's a discussion on this somewhere on this board, I recall, because I wrote the thing. :) It shows exact magnitude responses from 3 different filters, minimum, pure delay, and some other choice (in mixed phase FIR filters, you have 3 choices for every set of zeros corresponding to one system root, both inside, one each, and both outside. This is independent for EVERY set, so if you have a filter with 50 quads of roots, you have 3^50th possible impulse responses with exactly the same magnitude response.
- Mixed phase (in context of rooms) - another term for measured phase, i.e. loudspeaker's minimum phase response + excess phase.
Ok, modulo the "excess phase" thing. When you separate something into minimum phase (including zeros on the unit circle in minimum for practicality), the remaining roots are actually MAXIMUM PHASE. Yes, you can factor a filter into 3 parts convolved together, the minimum phase, delay, and maximum phase parts. That's a better way to put it, I think.

As alluded to earlier, there is also absolute phase and relative phase. Absolute phase = where phase integrity is maintained between left and right. For e.g. if you flip the polarity of both speakers, you won't hear it. Relative phase = where the phase of left and right is misaligned, for e.g. if you flip the polarity of one speaker you will definitely hear it. Sound seems to be coming from inside your head. If you do a sweep of both speakers playing together, you will see comb filtering. Having said that, phase and polarity are not the same thing, but this post is getting too long already.
So it's clear, we are talking mostly about INTRA-PHASE (single channel phase response) here. Yes, that's audible. Sometimes. Often it's not. "It depends" and the only way to know for sure is to actually examine its effect on the human cochlear analysis. Yeah, I didn't say simple.

It took me months to understand all this confusing terminology. It is misleading to say that all phase rotations are inaudible. You have to refer to exactly what type you are talking about.

Boy, howdy you can say that again, with feeling.
 

j_j

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If it’s inaudible, why worry about over correcting it? Why are we using FIR filters at all? If it’s inaudible, why aren’t FIR filters considered a solution in search of a problem at the expense of additional processing power and potentially needing to manage significant latency? I realize I’m missing something. Can someone possibly shed some light on what I’m missing? Please and thank you.

That was a lot of (justifiable) questions. I'm trimming out the "is it audible" with the comment "sometimes". (more so inter aural than intra-aural, but both can matter substantially).

The "why FIR" is simple, though. All FIR's are stable and can be created using reasonable processes. You can make bad filters, good filters, etc, but you can't make one that's unstable.

IIR's MUST be made with minimum-phase poles (denominators of the polynomial). Otherwise they go "boom", mathematically, not just practically.

Furthermore, FIR's may require much more length (you have to compare impulse response length to get to the bottom of it, IIR's can have long impulse response).
FIR's however, do not require gigantic numerical accuracy, and many times an IIR filter can require atrocious numerical accuracy (it's easy to design a filter at 48kHz that barely works in a double float. EASY, and some can require "the hell you say" length mantissa), or it will have an intrinsic noise floor higher in amplitude than the available mantissa. In such cases "stuff happens", what stuff is not easily calculated, even, but all the "stuff" is bad stuff.

On the other hand if you need a slow, broad pre-emphasis filter or something like that, the IIR may be the trivial solution.

So, it's application dependent.
 

j_j

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Link fixed.


That's for a minimum phase, "linear" phase, and maximum phase set of filters with identical magnitude response.


This might help understanding the minimum, delay, maximum phase issues.
 
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levimax

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Thank you @j_j , it is great to get good information on subjects like this where there is a lot of well meaning but often times incorrect information existing. I have a question about FIR filter taps. Is there a rule of thumb for how many are required in a crossover application? I know more are needed at low frequency vs high frequency but outside of loading the filters into Rephase and looking at the graph that shows the actual filter vs the target filter and adding or subtracting taps until they match closely is there a way to figure this out quickly and easily? Thanks
 

j_j

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Thank you @j_j , it is great to get good information on subjects like this where there is a lot of well meaning but often times incorrect information existing. I have a question about FIR filter taps. Is there a rule of thumb for how many are required in a crossover application? I know more are needed at low frequency vs high frequency but outside of loading the filters into Rephase and looking at the graph that shows the actual filter vs the target filter and adding or subtracting taps until they match closely is there a way to figure this out quickly and easily? Thanks

There are approximations in Rabiner and Gold. Because the issue of where zeros land as you change length, somewhat annoyingly, no, there is not really good approximation. All of edge of pass band, edge of stop band, inband ripple, and stop band ripple are included in the problem. This is when using things like "firpm" in matlab are useful, or "remez" in octave if you can get octave to run. (It's annoying)

Maybe I should start a thread on some of the complexities, and show some examples. That's probably overkill for this thread.
 

j_j

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There are approximations in Rabiner and Gold. Because the issue of where zeros land as you change length, somewhat annoyingly, no, there is not really good approximation. All of edge of pass band, edge of stop band, inband ripple, and stop band ripple are included in the problem. This is when using things like "firpm" in matlab are useful, or "remez" in octave if you can get octave to run. (It's annoying)

Maybe I should start a thread on some of the complexities, and show some examples. That's probably overkill for this thread.

I should add, for crossovers, one wants an "even order" filter, which means an odd number of taps. This provides an integer sample delay for the filter. This has very handy-dandy consequences. :)
 

gberchin

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Using long FIR filters to approximate the truncated impulse response of the unstable inverse of an allpass filter (compensation for the "non-minimum" phase portion of the response) can be tricky enough to qualify as an "art".
 

312elements

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I’m clearly out of my league here and I appreciate all of your contributions to the conversation. It’s become clear that I’m sitting at the wrong table but if you could humor one more question…

Let’s assume that I’ve got a well measuring 3 way speaker. A speaker that @amirm would deem “EQ-able”. I take my in room response measurement and there are a few areas that can be helped with equalization. Am I degrading the audio quality “fixing” those problems with PEQ because of the phase issues created from the filters or would it be more appropriate to use rePhase to create the filters to avoid creating audible phase distortion. I feel like you’ve answered this question if I were educated enough to read between the lines, but I’m still unsure about how to proceed and this seems like a specific enough scenario to avoid an “it’s complicated” answer.
 

levimax

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I’m clearly out of my league here and I appreciate all of your contributions to the conversation. It’s become clear that I’m sitting at the wrong table but if you could humor one more question…

Let’s assume that I’ve got a well measuring 3 way speaker. A speaker that @amirm would deem “EQ-able”. I take my in room response measurement and there are a few areas that can be helped with equalization. Am I degrading the audio quality “fixing” those problems with PEQ because of the phase issues created from the filters or would it be more appropriate to use rePhase to create the filters to avoid creating audible phase distortion. I feel like you’ve answered this q anduestion if I were educated enough to read between the lines, but I’m still unsure about how to proceed and this seems like a specific enough scenario to avoid an “it’s complicated” answer.
FR issues are much more audible than phase issues (outside of some pathologic edge cases) so proper room EQ is going to be a net positive. Since there are free tools you can try EQ for your self with and without phase corrections.
 

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I’m clearly out of my league here and I appreciate all of your contributions to the conversation. It’s become clear that I’m sitting at the wrong table but if you could humor one more question…

Let’s assume that I’ve got a well measuring 3 way speaker. A speaker that @amirm would deem “EQ-able”. I take my in room response measurement and there are a few areas that can be helped with equalization. Am I degrading the audio quality “fixing” those problems with PEQ because of the phase issues created from the filters or would it be more appropriate to use rePhase to create the filters to avoid creating audible phase distortion. I feel like you’ve answered this question if I were educated enough to read between the lines, but I’m still unsure about how to proceed and this seems like a specific enough scenario to avoid an “it’s complicated” answer.
The phase shift that occurs and is present from capacitive reactance and inductive reactance is not avoidable. Using PEQ helps significantly even if some additional phase shift occurs in a PEQ operation. Decades ago I calculated the phase shifts from using analogue filters and determined that it was there but it was not significant enough to be audible. It is impossible to avoid phase shift so my best advice is to relax and enjoy the advantage that PEQ affords the system and listen to the major improvement heard in the music. The crossovers and the speaker drivers have a lot of reactance and introduce a lot of phase shift so a little more from PEQ makes little to no audible difference.
 

Keith_W

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Thank you for your explanations @KSTR and @j_j.

Lately I have been experimenting with phase correction with my DSP setup to decide for myself what is audible and what isn't. As an experiment I did an absolutely crazy correction to see what would happen. I took a nearfield measurement of each driver and looked at the phase response. Instead of linearizing it with a reverse AP filter, I simply inverted the phase and convolved that into the driver filter. The result was a phase curve so straight and flat that you could place a spirit level on top of it. This had the effect of really messing up the timing, so I had to time align the drivers with crazy amounts of delay. I then generated the rest of the filters with the same target curve to remove tonality as a variable, and had a listen.

It was absolutely dreadful. Not subtle at all. The "Control" (without any phase correction) had an open, natural sound. The version with my stupid phase filter experiment sounded constricted, with the soundstage lifted off the floor and squashed down from the ceiling so it sounds as if you are listening to music through a narrow horizontal slit in a pillbox. I have no idea why left/right speaker correction should squish the vertical plane. Still, it was a worthwhile experiment. Reading descriptions on the internet is no match for actually trying it yourself and experiencing the effect. Experts might say "don't do this" - I can appreciate that on an abstract level, but nothing beats experiencing the reality of why you should not do this for yourself.

I have done other experiments of course, each time making sure that all the other known variables that affect sound - tonality, amplitude cancellations, time alignment, etc. are kept the same. I only mentioned the experiment with the worst outcome. In fact, all my experiments so far have resulted in worse outcomes compared to leaving the damned thing alone and doing no phase corrections. But I am not ready to draw the conclusion that I should leave it alone yet. There are many reasons why I am getting bad results, and the biggest reason is lack of knowledge of what to correct and how.
 

Doodski

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Thank you for your explanations @KSTR and @j_j.

Lately I have been experimenting with phase correction with my DSP setup to decide for myself what is audible and what isn't. As an experiment I did an absolutely crazy correction to see what would happen. I took a nearfield measurement of each driver and looked at the phase response. Instead of linearizing it with a reverse AP filter, I simply inverted the phase and convolved that into the driver filter. The result was a phase curve so straight and flat that you could place a spirit level on top of it. This had the effect of really messing up the timing, so I had to time align the drivers with crazy amounts of delay. I then generated the rest of the filters with the same target curve to remove tonality as a variable, and had a listen.

It was absolutely dreadful. Not subtle at all. The "Control" (without any phase correction) had an open, natural sound. The version with my stupid phase filter experiment sounded constricted, with the soundstage lifted off the floor and squashed down from the ceiling so it sounds as if you are listening to music through a narrow horizontal slit in a pillbox. I have no idea why left/right speaker correction should squish the vertical plane. Still, it was a worthwhile experiment. Reading descriptions on the internet is no match for actually trying it yourself and experiencing the effect. Experts might say "don't do this" - I can appreciate that on an abstract level, but nothing beats experiencing the reality of why you should not do this for yourself.

I have done other experiments of course, each time making sure that all the other known variables that affect sound - tonality, amplitude cancellations, time alignment, etc. are kept the same. I only mentioned the experiment with the worst outcome. In fact, all my experiments so far have resulted in worse outcomes compared to leaving the damned thing alone and doing no phase corrections. But I am not ready to draw the conclusion that I should leave it alone yet. There are many reasons why I am getting bad results, and the biggest reason is lack of knowledge of what to correct and how.
What are you using?
 

j_j

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Thank you for your explanations @KSTR and @j_j.

Lately I have been experimenting with phase correction with my DSP setup to decide for myself what is audible and what isn't. As an experiment I did an absolutely crazy correction to see what would happen. I took a nearfield measurement of each driver and looked at the phase response. Instead of linearizing it with a reverse AP filter, I simply inverted the phase and convolved that into the driver filter. The result was a phase curve so straight and flat that you could place a spirit level on top of it. This had the effect of really messing up the timing, so I had to time align the drivers with crazy amounts of delay. I then generated the rest of the filters with the same target curve to remove tonality as a variable, and had a listen.

It was absolutely dreadful. Not subtle at all. The "Control" (without any phase correction) had an open, natural sound. The version with my stupid phase filter experiment sounded constricted, with the soundstage lifted off the floor and squashed down from the ceiling so it sounds as if you are listening to music through a narrow horizontal slit in a pillbox. I have no idea why left/right speaker correction should squish the vertical plane. Still, it was a worthwhile experiment. Reading descriptions on the internet is no match for actually trying it yourself and experiencing the effect. Experts might say "don't do this" - I can appreciate that on an abstract level, but nothing beats experiencing the reality of why you should not do this for yourself.

I have done other experiments of course, each time making sure that all the other known variables that affect sound - tonality, amplitude cancellations, time alignment, etc. are kept the same. I only mentioned the experiment with the worst outcome. In fact, all my experiments so far have resulted in worse outcomes compared to leaving the damned thing alone and doing no phase corrections. But I am not ready to draw the conclusion that I should leave it alone yet. There are many reasons why I am getting bad results, and the biggest reason is lack of knowledge of what to correct and how.

Uh, yeah, you left out some stuff, like what crossover you have and how it works. Also if you reversed the phase, did you also invert the driver response so as to flatten it, rather than simply flop the phase. If so you may have actually emphasized the frequency response errors. Without a system diagram, I can't really be sure what you did.
 

Keith_W

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I am using a DSP controlled 8 way active system using filters made with Acourate. The crossovers are lin phase FIR's. And yes, I did flatten the driver response and the phase.

1713239247095.png


This is the woofer measurement before (red) and after (green), prior to convolution with the bandpass filter.

Like I said, the result was really satisfying to see. But listening to the result was a different matter entirely.
 

René - Acculution.com

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While Pauls video is too hand-wavy and mixes up phase and time, I also don't think Amir's video has the proper nuance, as the peer-reviewed papers probably don't tell us the full story. As this thread demonstrates, lots of people have experience with phase differences relating to auditory differences, and someone like @KSTR ;-) should really do a video summing up both theory and with listening examples. One complication is that when you loudspeaker has already messed up the phase, differences are probably more difficult to hear, which ties in with why the peer-reviewed papers mention a difference between headphones and loudspeakers, and so for many people it will be difficult to compare different phase characteristics on their HiFi, as they don't have a good starting point.
 

mdsimon2

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I am using a DSP controlled 8 way active system using filters made with Acourate. The crossovers are lin phase FIR's. And yes, I did flatten the driver response and the phase.

View attachment 364057

This is the woofer measurement before (red) and after (green), prior to convolution with the bandpass filter.

Like I said, the result was really satisfying to see. But listening to the result was a different matter entirely.

A few things stick out to me. Your corrected magnitude trace has high pass behavior but your phase trace is essentially flat. This is what @KSTR was referring to in the quote below in response to @Tim Link.

At the moment you have a "dirty" measurement-derived correction and notably you are overcorrecting the low bass to linear-phase, not following the natural phase rotation of the highpass function. That is guaranteed to give pre-ringing/-signal in the lowest bass.

As a result your correction has non-minimum phase behavior baked in. If correcting a single driver, you should be able to flatten phase and magnitude response with only minimum phase corrections.

Correct me if I am wrong but I believe in other threads you mention that your approach is to make near field measurements for all drivers, including high frequency drivers. I also do not think you add baffle effects to your near field measurement. Near field measurements will not be accurate for high frequency drivers and without considering baffle effects your woofer response will also be inaccurate. All of this makes any driver correction questionable. You really need to start with quasi-anechoic measurements if attempting to correct at the speaker level.

Michael
 

gnarly

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But I am not ready to draw the conclusion that I should leave it alone yet. There are many reasons why I am getting bad results, and the biggest reason is lack of knowledge of what to correct and how.

Yeppers. That's where I stand with room corrections. (I feel I have speaker building, at the anechoic level, figured out pretty well.)

Speakers in rooms???
What to correct and how? ....along with equally important knowing what cannot be corrected....is the name of the game, imo.
I'd much rather have that complete knowledge set than any particular DSP, measurement program, or room correction / filter generating software.

Lee Trevino, a famous past golfer, was asked by a reporter how much the top-of-line clubs given to touring professionals for endorsements, contributed to the pros being able to win....
He quipped, "A pro will beat you with an umbrella"

I need a good course in Room Corrections Made with an Umbrella LOL
 
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