The Nautaloss Ref Monitor

Also tested were some Linn and Totem speakers. Cool. Glad there is no issue. I am off to figuring it out on my end. BTW, there are some nasty 3rd hamronic spikes in the lower part of the vocal range as well as a rise in distortion in LF, but I cant say that it is a visual representation of what we were hearing.
 
I see the distortion spikes you are talking about at 230 Hz - but that is still -37dB down - well below 0.1% - which I think is below the point of being perceptible. Down below 100 Hz, I expect the distortion to get worse in general as that is where the XO was applied. I should have done a straight bypass switch within miniDSP just to see what the ADC and DAC chain does to the signal.
 
I don't see why you say it's difficult to make any conclusions. The room effect is identical on both because the mic and speaker were untouched. The only change was how the signal to the amp was routed. Besides, room effects are below 500 Hz and the complaint from the member who suggested this test said that a HF 'veil' was present when the miniDSP is in the path. Clearly, the measurements show a perfect match on the frequency response from 200 Hz up to 20 kHz - the only difference being an improved phase linearity at expense of impulse response overshoot. From this, I can conclude from this measurement and my own ears that the miniDSP causes no noticeable degradation in the HF sound quality associated with the key frequency band associated with imaging, localization, and "airiness" as some people call it. Definitely no sign of a HF "veil" and if anything, the impulse overshoot may be heard (if you can detect it audibly) as a sharpness in percussion instruments - just the opposite of a veil.

I don't have an ADC input that can measure amplifier voltage transients and waveforms (an O-scope basically) and don't think it will give me any more info than what I am looking for here - measurable acoustic differences at the speaker output.

U have more tools at your disposal than you thought 🙂

Oscilloscope - Software: Oscilloscope
Soundcard Scope
Winscope
http://www.diyaudio.com/forums/pc-b...ope-spectrum-analyzer-function-generator.html
 
xrk, a man on a mission, cool.

Don't have much to offer on this topic other than a couple recent encounters with the "less known"

Recently a friend and I built a system in his new, amazing 😱 21x41 garage...a true man cave. The goal, fill the space with "big" sound that isn't a train wreck.

Proceeding, 2 coaxial Beta 10's with CTS Piezo's and a pair of BFM folded horns with 12" Kappa lites were built I ran the coax's at home for a quick taste which confirmed we were on the right track, plenty of SPL and no eminent collision... yet.

The difficult part came during the installed. Way too many hard, reflective surfaces! (I think corner, floor to ceiling, line arrays would have been better but that didn't work with his layout). So, using a Mini DSP I eq'd the heck out of the system and got a little flatter response curve. The down side...lost a whole bunch of power and the sound...well...lets just say it wasn't a wreck but a derailment. The next steps were to put the mic away and dialed the eq'g way back. We had power again and it sounded... really live, again. Best part of this story was, my friend loved the big room filling sound, reverb and all. In his words "it sounds like a live band is right here...in the garage with us. Hemmm, a garage band.

Second encounter was with a set of Frugal Horns and a DCX 2496. I ran the FH's wide open, no filtering or crossing, amp'd outside the 2496, then integrate this unit, and amps, to cross a tweeter in around 9k and a sub in at about 65hz. This actually work rather well. Slightly crisper up top and a whole lot more bass presence/pressure on the bottom. My conclusion here was that while this combination presented an interesting new sound it ended up taking something away from the pure, simple, natural sound of these speakers.

In the end I came away with two things. one, good designs that do what they are intended for should be enjoyed for just that... and two, heavy eq'g is not a cure all for acoustic space challenges. Darn.

So, for what it's worth, there's a couple crude, eq'g stories along this apprentices journey.

Hope you make it a music day 🙂
Marko
 
Marko,
Your stories ring very true and I will say that for the fullrange driver, I only EQ the very top to help the HF falloff. For the sub, I don't think it is a big problem to EQ the bass flat below 150 Hz as long as you don't hit xmax. I don't touch anything between 400 Hz and 7kHz. I agree that simple full range speakers like my wall mounted BIB's still sound very nice and have a presence and naturalness that is a hallmark of a good full range speaker. I also enjoy my Cornu's and Karlsonators very much still.
 
Without going into the theory/technical discussion, I can attest that sparse gentle use of EQ can be more helpful than harmful... It's done in mixing and mastering everyday. EQ'ing for a room is not so different, but you have to use your ears and not just aim for a "flat graph".

Most people's first attempt at EQ'ng a room involves using lots of filters to achieve a perfectly flat response (and this will smear the material badly).
But don't give up, your room has a difficult response that could benefit from at least 1 band of EQ (and probably not so many more bands than that 🙂.

Because there is no achievable 'ideal', system tuning is a delicate art of balancing compromises. Using as little EQ as possible is a good start since every pole causes phase shift *or* pre-ringing (you choose). One technique mastering/mixing engineers use is to cut instead of boost. In the case of minimum phase filters this puts the phase shift at the bottom of the cut (in other words there is less energy where the phase shift is occurring making it less noticeable). This rule can be summarized as "don't boost valleys, just cut peaks".

Another longstanding technique is to use broad curves... Visually average the energy in a region and instead of targeting the peaks and valleys individually use a broad stroke to equalize it's overall cumulative response. This will result in a near invisible correction (but which won't perfectly correct the problem -> keep juggling). Also, knowing how the ear responds to specific frequencies is vital to consider when applying EQ because you can factor in sensitivity to phase & power.

It's beyond this discussion, but I strongly recommend broadband room treatment and bass trapping before EQ. My approach has been to take care of whatever response anomalies exist physically before applying DSP processes (analog system's artifacts are more forgiving/entropic than digital systems). The results of good room treatment + delicate corrective DSP can be amazing. It's not the only way forward but it can produce a great experience.

Footnote, Having owned and used many mastering grade converters along with all manner of integrated codec devices I can IME say ADC/DAC quality is only *sometimes* noticeable on a good system. I would prioritize conversion after:
1.) room treatment
2.) speakers
3.) amplification
Of course, putting pressure on miniDSP developers to use a near TOTL chipset seems reasonable. For now we can use the miniDSP via i2s and avoid double conversion.
 
Good stuff Anthony,
I want to be careful and not derail xrk's thread into an eq'g discussion, that' his really his prerogative, but I'd like to get a better understanding of phase shifting. I'm familiar with the basic's... but not with behavior... if you will. An suggested 101 threads or other reads on this would be helpful.
Thanks
 
Here is the amp:
miniAMP | MiniDSP

Here is digital I/O board:
miniDIGI | MiniDSP

I think both of these can stack on the standard 2x4 miniDSP - you just won't use the analog RCA in/out that comes with it.

There is also the Curryman DAC that seems to be popular:
Curryman DAC (ES9023) | MiniDSP

I think to implement FIR filters (linear phase) the miniSHARC + Curryman DACs are the way to go:
https://www.minidsp.com/products/opendrc-series/minisharc-kit

I think for a 2x4 XO with EQ you will need the digital I/O board (unless you have I2S already), the minSHARC, and qnty 2 Curryman DACs. You are looking at $325 total but it will be powerful real time FIR capable system. Add more Curryman DACs for more channels I think.

Best to ask minDSP tech support, I still don't have a firm grasp of what to get with what.

Nothing wrong with class D. In fact, it rocks! Check out TPA3116D2 amp in class D forum.
 
Definetely is not the most clearly explained site. I am a Class A, minimal circuitry guy. You want an awesome amp that is not Class A(can be biased that way as option), would be relatively cheap, but blast Class d, check out the TSSA V1.7.
 
Without going into the theory/technical discussion, I can attest that sparse gentle use of EQ can be more helpful than harmful... It's done in mixing and mastering everyday. EQ'ing for a room is not so different, but you have to use your ears and not just aim for a "flat graph".

Most people's first attempt at EQ'ng a room involves using lots of filters to achieve a perfectly flat response (and this will smear the material badly).
But don't give up, your room has a difficult response that could benefit from at least 1 band of EQ (and probably not so many more bands than that 🙂.

Because there is no achievable 'ideal', system tuning is a delicate art of balancing compromises. Using as little EQ as possible is a good start since every pole causes phase shift *or* pre-ringing (you choose). One technique mastering/mixing engineers use is to cut instead of boost. In the case of minimum phase filters this puts the phase shift at the bottom of the cut (in other words there is less energy where the phase shift is occurring making it less noticeable). This rule can be summarized as "don't boost valleys, just cut peaks".

Another longstanding technique is to use broad curves... Visually average the energy in a region and instead of targeting the peaks and valleys individually use a broad stroke to equalize it's overall cumulative response. This will result in a near invisible correction (but which won't perfectly correct the problem -> keep juggling). Also, knowing how the ear responds to specific frequencies is vital to consider when applying EQ because you can factor in sensitivity to phase & power.

It's beyond this discussion, but I strongly recommend broadband room treatment and bass trapping before EQ. My approach has been to take care of whatever response anomalies exist physically before applying DSP processes (analog system's artifacts are more forgiving/entropic than digital systems). The results of good room treatment + delicate corrective DSP can be amazing. It's not the only way forward but it can produce a great experience.

Footnote, Having owned and used many mastering grade converters along with all manner of integrated codec devices I can IME say ADC/DAC quality is only *sometimes* noticeable on a good system. I would prioritize conversion after:
1.) room treatment
2.) speakers
3.) amplification
Of course, putting pressure on miniDSP developers to use a near TOTL chipset seems reasonable. For now we can use the miniDSP via i2s and avoid double conversion.

If one uses the EQ built in to iTunes or some other compressed playback program does that mess with the phase? What about EQ at the digital file level say when you are editing/mixing in a nonlinear editing program like Audacity and you apply an EQ then render the track to flac or mp3 when that is played back, did the EQ'ing mess with the phase?
 
Absolut not an expert, but I know JRiver mediacenter can do FIR filters and XO. So as trial one can use Jriver free 30 days period to exercise. To make the filters see link for REPHASE. Very interesting this REPHASE thread but takes times to investigate http://www.diyaudio.com/forums/mult...zation-eq-fir-filtering-tool.html#post3199084. At REPHASE thread I get the sense that running the filters on a desktop is soundwise fine, but the miniShark wins when we talks compute time delay.