New EX MACHINA ARCTURUS 3-WAY ACTIVE STUDIO MONITORS

[PowerSerge]

Im sure they sound amazing but they look diy

 

[ernestcarl]

Right, whatever it's called I would definitely not want something like that for monitoring. 45ms is pretty hardcore long latency as well, for studio work it means it's impossible to play on any electric/electronic instrument while listening through them, that's the reason there's low latency mode and you have to manually switch between them

Some Adam subwoofers have even longer processing delays for phase correction, but they do not linearize the phase of the speakers to the extreme extent Ex Machina does. Yeah, the low latency mode would be the default for most live monitoring workflows.

 

[Tim Link]

Erin reviewed the smaller Pulsar which yielded the ff. step response:

View attachment 306215

Ex Machina Pulsar MKII 3-way 8-inch Active Studio Monitor Review

Ex Machina Pulsar MKII Review

www.erinsaudiocorner.com

The low end might not exactly audibly "pre-ring"...

@KSTR briefly had this to say across at DIYAudio:

"... this appears to be a speaker where the natural phase response from the woofer highpass was corrected to linear phase which creates the point-symmetric step response.
That's a neat trick for speakers to have "fast bass" down low. The price you pay is that some signals (like certain kick drum sounds) may be rendered with audible artifacts. The rising slope in the time-domain is not ringing and does not sound like ringing, rather it sounds like pink noise pulse ramping up (very interesting acoustical illusion)."

Why would that pink noise pulse ramping up effect be? Is this true for electrical devices as well, and is this effect only audible in the lower bass, or would any driver need to have an associated group delay with it's LF roll off to avoid this? Would the effect disappear or get shifted to below audible if a subwoofer was added to this speaker and integrated so that it went down to 5 Hz with the phase still flat? I'm really curious about how and why this effect would happen. Must it always happen? Is it intrinsic to the nature of a signal with flat phase through the high pass zone?

 

[dfuller]

Why would that pink noise pulse ramping up effect be? Is this true for electrical devices as well, and is this effect only audible in the lower bass, or would any driver need to have an associated group delay with it's LF roll off to avoid this? Would the effect disappear or get shifted to below audible if a subwoofer was added to this speaker and integrated so that it went down to 5 Hz with the phase still flat? I'm really curious about how and why this effect would happen. Must it always happen? Is it intrinsic to the nature of a signal with flat phase through the high pass zone?

Okay, so TLDR: it's because correcting phase response to linear for the bass doesn't make sense. The linear phase is not worth the bodies that it buries.

The reason it works for crossovers and not for the low end roll off is because a HPF and LPF will "correct" for one another's phase response. A typical minimum phase (read: passives, or actives sans phase correction) crossover has phase wraps around the crossover point(s) from the pass filters. They will each get to a point where you'll see, basically, inverse phase rotation for each filter - and that can be corrected for pretty easily via all-pass filters (analog, a la PSI) or in DSP with FIR filters. This doesn't cause pre-ringing, even with super steep filters (8th order, even, is OK).

But it doesn't work for LF roll off because there's no LPF to go along with the HPF of the system, so instead you just get that weird ramping effect.

If you add a sub, you now have a LPF to work with.

 

[ernestcarl]

Why would that pink noise pulse ramping up effect be? Is this true for electrical devices as well, and is this effect only audible in the lower bass, or would any driver need to have an associated group delay with it's LF roll off to avoid this? Would the effect disappear or get shifted to below audible if a subwoofer was added to this speaker and integrated so that it went down to 5 Hz with the phase still flat? I'm really curious about how and why this effect would happen. Must it always happen? Is it intrinsic to the nature of a signal with flat phase through the high pass zone?

"Electrically", I don't think the same issue applies with the kind of time smearing side-effect seen with physical transducers.

It appears we are more sensitive to precausal/acausal effects at low frequencies -- papers/studies were posted elsewhere before... As dfuller stated, since there is no xo distortion to correct below our subwoofer there's really little point to completely linearize the phase all the way down that low.

I think if the "correction" is mild enough, artifacts if there's any to be seen may be low enough in level not to be a noticeable audible detriment. Although, I also do agree that it seems more hassle than it's worth in many situations.

 

[ernestcarl]

Some Adam subwoofers have even longer processing delays for phase correction, but they do not linearize the phase of the speakers to the extreme extent Ex Machina does. Yeah, the low latency mode would be the default for most live monitoring workflows.

Hedd not Adam. I confused the two.

 

[Tim Link]

Okay, so TLDR: it's because correcting phase response to linear for the bass doesn't make sense. The linear phase is not worth the bodies that it buries.

The reason it works for crossovers and not for the low end roll off is because a HPF and LPF will "correct" for one another's phase response. A typical minimum phase (read: passives, or actives sans phase correction) crossover has phase wraps around the crossover point(s) from the pass filters. They will each get to a point where you'll see, basically, inverse phase rotation for each filter - and that can be corrected for pretty easily via all-pass filters (analog, a la PSI) or in DSP with FIR filters. This doesn't cause pre-ringing, even with super steep filters (8th order, even, is OK).

But it doesn't work for LF roll off because there's no LPF to go along with the HPF of the system, so instead you just get that weird ramping effect.

If you add a sub, you now have a LPF to work with.

So the gist is we can't really hear the group delay down there, but we might hear the ramp up effect some times if we try to force the phase flat. I wish I understood how that works better. I should just make linear phase filters with steep cutoff rates at different frequencies and listen to music through them on headphones. See if I can hear that effect. Headphones should do it was well as speakers. In any case, it seems the phase can be flattened to a significant degree without noticeable effect, and without too much effort.

 

[dfuller]

So the gist is we can't really hear the group delay down there, but we might hear the ramp up effect some times if we try to force the phase flat. I wish I understood how that works better. I should just make linear phase filters with steep cutoff rates at different frequencies and listen to music through them on headphones. See if I can hear that effect. Headphones should do it was well as speakers. In any case, it seems the phase can be flattened to a significant degree without noticeable effect, and without too much effort.

Effectively, yes - you're hearing the pre-ring from a linear phase HPF. And yeah, headphones make it more obvious but any decent speaker should demonstrate it clearly enough.

 

[gnarly]

So the gist is we can't really hear the group delay down there, but we might hear the ramp up effect some times if we try to force the phase flat. I wish I understood how that works better. I should just make linear phase filters with steep cutoff rates at different frequencies and listen to music through them on headphones. See if I can hear that effect. Headphones should do it was well as speakers. In any case, it seems the phase can be flattened to a significant degree without noticeable effect, and without too much effort.

I dunno ....about not being able to hear bottom-end group delay.

The prevailing opinion seems to be sealed subs sound better than vented, due to their lower order roll-off.
I think most of that opinion is probably formed for wrong reasons, mainly due to the difficulty of actually getting sealed subs to extend as low as vented with equal power, and no damn low corner boost. (but that's a whole 'nuther can of worms)

i think when the rather comparatively massive sealed displacement it takes to match vented at f-3 exists, it supports the opinion sealed sounds better.
Which to me, means group delay can be heard.

That said, I haven't had any luck with headphone tests of bottom end group delay. It should be easy with headphones, to just use all-pass, but I dunno...i just don't ever hear much difference.
Whereas with speakers outdoors, I can hear the difference in a bank of sealed subs vs same output capability using vented.
Difference is in big low end transients, that move a lot of air. The tightness of the hit.

Anyway, I guess I'm sayin if you want to explore phase flattening of low end roll-off, ime/imo it will be best with a speaker.

Either way, it will take a good number of taps. I think maybe 16k taps @ 48kHz sample rate is probably a good minimum choice. Frequency resolution for that is 6Hz.
A decent rule of thumb seems to be have freq resolution at least 1/3 of lowest freq intended to correct. So 16k @ 48 is probably good for work down to 18Hz.
Oh, and as a reminder, if the sample rate used is 96kHz, it takes twice the taps for equivalence.

The only times I've felt confident I can consistently hear pre-echo, has been with more taps and also using then to effect a linear-phase system high-pass.
Like 65k taps at 48kHz. You may want to play with that too.

Good luck with experiments...look forward to hearing your impressions.

 

[Tim Link]

Either way, it will take a good number of taps. I think maybe 16k taps @ 48kHz sample rate is probably a good minimum choice. Frequency resolution for that is 6Hz.
A decent rule of thumb seems to be have freq resolution at least 1/3 of lowest freq intended to correct. So 16k @ 48 is probably good for work down to 18Hz.
Oh, and as a reminder, if the sample rate used is 96kHz, it takes twice the taps for equivalence.

The only times I've felt confident I can consistently hear pre-echo, has been with more taps and also using then to effect a linear-phase system high-pass.
Like 65k taps at 48kHz. You may want to play with that too.

Good luck with experiments...look forward to hearing your impressions.

How do you calculate low frequency resolultion based on taps and sample rate? Also, why do more taps make the pre-echo more audible? I would normally expect that more is better.

 

[gnarly]

Highly recommend SynAud Cons articles on FIR:

https://www.prosoundtraining.com/2013/09/06/fir-ward-thinking-fir-filters-audio-systems/

https://www.prosoundtraining.com/2013/09/27/fir-ward-thinking-part-2/

Here's a snip from part3 of that series, which can be found at:
https://www.prosoundweb.com/fir-ward-thinking-continued/

I think the reason I heard pre-echo with more taps is because of incorrect implementation. That being having a linear-phase system high-pass.
More taps simply illuminated the incorrectness.

Otherwise more taps should just give better frequency resolution. But again, that can be a two edged sword if FIR is incorrectly implemented, providing overcorrections for example.
I see less taps as a form of "correction smoothing" sometimes.

 

[Tim Link]

I see less taps as a form of "correction smoothing" sometimes.

Is that different than smoothing and windowing the response before applying correction? I've been shooting for a smooth response at the listening position with about 7 cycle windowing lately. The reason is that when I finally looked at the windowed response at the listening position I was seeing big drops in the frequency response at the crossovers, which didn't appear with longer windowing. I ruled out interference problems by measuring each driver separately. The nearfield equalization in my case wasn't holding out to the listening position, with more droop in the low end as I moved away from my tweeters. This has made a huge improvement in my perception of the sound quality.

Since I'm trying to correct at distance I know I'm still letting in some of the room at that point so I smooth too. The worst thing I've heard from FIR filters so far is from giving them too much information to work with. My first attempts at room correction were to window at 35 cycles with no smoothing, based on an OCA video I watched. That wasn't good! I may have missed a step in his process, but for sure you will hear strange things if you do that. Better to under detail than to over detail the corrections.

Since I started making sure I'm not making any sharp corrections I haven't heard any obnoxious artifacts. My rule is that the correction curve should not have any hard kinks in frequency response or phase response in it.

 

[Tim Link]

Either way, it will take a good number of taps. I think maybe 16k taps @ 48kHz sample rate is probably a good minimum choice. Frequency resolution for that is 6Hz.
A decent rule of thumb seems to be have freq resolution at least 1/3 of lowest freq intended to correct. So 16k @ 48 is probably good for work down to 18Hz.
Oh, and as a reminder, if the sample rate used is 96kHz, it takes twice the taps for equivalence.


Good luck with experiments...look forward to hearing your impressions.

I don't know how to determine how many taps my filters have. If I export an impulse from REW, is there an export setting that tells me the number of taps that will be employed? Or is that a setting on the convolution engine? I see I can choose the sample count and the sample rate on the impulse export.

 

[gnarly]

I've only played with using REW as a FIR generator briefly, and in the past... https://www.diyaudio.com/community/threads/rew-as-fir-generator-experiments.349880/
As I remember, the key was setting Left and Right width times to correctly reflect desired sample counts in "IR Windows". So it just took a little sample to ms conversion math.

I do not know what changes have been made to REW since that thread from 2020. I see the sample rate section in Impulse Export along with a number of options.
I'm pretty sure Apply IR Window before Export, and then getting the rest to match will be the key.
But dunno, and don't really want to explore using REW for FIR anymore.....too easy to screw up, i think.

 

[Tim Link]

But dunno, and don't really want to explore using REW for FIR anymore.....too easy to screw up, i think.

Screw up as in fail to fully optimize, or screw up as in create something really horrible?

 

[gnarly]

or screw up as in create something really horrible?

Let's just say I'd make it a mandatory requirement to run an electrical transfer function of the FIR filter, before I'd send it to an amp & speaker.
Levels can be tricky using trace arithmetic and impulse inversions......but that may just be my lack of practice with REW.

 

[terryforsythe]

Let's just say I'd make it a mandatory requirement to run an electrical transfer function of the FIR filter, before I'd send it to an amp & speaker.
Levels can be tricky using trace arithmetic and impulse inversions......but that may just be my lack of practice with REW.

I use REW to generate .wav convolution files to tune to the Harman curve, but with the bass boost reduced by about 4dB. I don't know whether it is perfect, but I haven't had any issues - it definitely improves the sound of my speakers' in-room response.

You can specify the number of taps when exporting to the .wav files, but yesterday I left that box unchecked and let REW choose.

I follow this tutorial:

EDIT: I made two changes to the operations when I generated my latest curves yesterday. One change I made from the tutorial is to uncheck "exclude notches" when that box pops up during the "Trace arithmetic" operations to generate the convolution trace. The other change was that I allowed a maximum of 10 dB of gain rather than cutting off everything over 0 dB.

 

[terryforsythe]

I don't know how to determine how many taps my filters have. If I export an impulse from REW, is there an export setting that tells me the number of taps that will be employed? Or is that a setting on the convolution engine? I see I can choose the sample count and the sample rate on the impulse export.

When exporting to a .wav file, you can specify the number of taps, or you can leave that box unchecked and let REW decide. On some convolution files I generated a few months ago I had specified 64k. Yesterday I unchecked the box and let REW decide the number of taps, and it seems to be working great.

 

[Tim Link]

When exporting to a .wav file, you can specify the number of taps, or you can leave that box unchecked and let REW decide. On some convolution files I generated a few months ago I had specified 64k. Yesterday I unchecked the box and let REW decide the number of taps, and it seems to be working great.

So 64K samples means 64K taps? That's a lot more than I usually see specified. I've been leaving that box checked. Latency doesn't seem too bad. I'm going to have to watch OCA's video again. That video is what started me down this whole FIR filter path.

 

[ernestcarl]

So 64K samples means 64K taps?

64k sample count export in REW appears equivalent to 65536 taps. Re-import the exported file into REW and look at the sample count indicated in the measurement notes.