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MQA, DSP and "sound quality"

Rodney Gold

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So you put two subs next to each other on each side wall? Or stacked them?
Effectively you did not distribute the bass sources, just two points?
Which Duke’s setup method you used? Can you link to Audiokinesis site?
I use his Swarm.
I had one midway all walls ... didnt stack them , ie mid front , back , side1 , side 2 .. my main speakers have 2 side firing woofers each .. so essentially 8 bass sources
Im pretty sure he had setup instructions somewhere or it might have been in a review or write up. The site is changed since I looked at it last

I dont use a swarm now ..albeit it did what it said on the tin , the subjective bass was not as good as the speakers alone and the upgrade to G1 spirits produces even better bass
 

oivavoi

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I am using Audiokinesis Swarm. Normally it comes with one mono Dayton sub amp, I got two for flexibility. Theoretically up to 80Hz there is no need for stereo.Some people claim to be able to locate sounds from 60Hz. That’s interesting given that almost all music has lows mixed as mono. In my setup mains work full range and provide all stereo clues. The four subs are scattered along the walls in a random manner. I tested them up to 200Hz LPF both in mono and stereo. In my experience the sources cannot be located up to 130-150Hz so mono is enough. If your mains work full range, mind you. Distributed bass is amazing. No need for EQ in my opinion.

Cool, thanks. The point for me is also to relieve the mains of their heavy lifting. I've been thinking of crossing over to distributed bass in mono from 100 hz. But I haven't yet heard such a setup, so I don't know how it will be.
 

Rodney Gold

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I've always felt this way, too, but I am beginning to look at it another way. If the purpose of the multiple spaced measurement mic placements is to supply enough information to create a relatively uniform response in the volume of space that they encompass. Now, if one measures anywhere in that space with any number of positions, the result will probably not replicate the "predicted" result but it should show a significant amelioration when A/B-ed with EQ bypassed. If it doesn't show that, the process failed.

Yep , same as the trinnov , it will correct for its measured space .. however defining a large space to correct doesnt do it as well as for a sweet spot
I have tried dirac in a single measurement scenario vs the 9 sweeps and its considerably better at the sweet spot than using multiples.
DIRAC have an upgrade in the pipeline soon , evidently its been fettled with etc .. all dirac licences get the new program when released.
 
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pirad

pirad

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Yep , same as the trinnov , it will correct for its measured space .. however defining a large space to correct doesnt do it as well as for a sweet spot
I have tried dirac in a single measurement scenario vs the 9 sweeps and its considerably better at the sweet spot than using multiples.
DIRAC have an upgrade in the pipeline soon , evidently its been fettled with etc .. all dirac licences get the new program when released.

Dirac does not advise single measurement in any scenario. Even the armchair position requires 9. Single measurement effectively deprives Dirac of its mixed-phase feature, their unique selling point.
Trinnov measures differently thanks to the special microphone.
 

Burning Sounds

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Well, I have tested both Acourate and Audiolense simulations, both in the frequency and time domains, which were "identical" using REW as the third party measurement software to validate. The "proof" is in the articles I wrote on CA for both products. I have extensively tested Acourate in my book, even to the point of measuring in REW, exporting the measurement, importing into Acourate and overlaying the simulation with the actual measurement - virtually identical. Same goes vice versa, and certainly within in the technical uncertainty of these acoustic measurement programs.

Yes, I've done the same with Acourate, but by your "vice versa" method - exporting the simulation into REW and then comparing it to the REW measurement - they are as you say, virtually identical.

It's also easy to drop the filter you have created into Acourate and re-measure and compare that to the simulation.

@mitchco if you don't mind answering a question here - is it possible to use Acourate for correction below Schroeder frequency only? I don't see a way to do it, but perhaps it can be done using the manual controls rather than the macros?
 
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pirad

pirad

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Cool, thanks. The point for me is also to relieve the mains of their heavy lifting. I've been thinking of crossing over to distributed bass in mono from 100 hz. But I haven't yet heard such a setup, so I don't know how it will be.
My Swarm XO is usually at 100Hz in mono. No problems whatsoever. But my mains run full range. I tried them with high pass filter too and that works fine, but I prefer not to put
any additional processing in the mains signal path. My speakers' midwoofer actually has no crossover whatsoever. Experiment, experiment, experiment as in Cole Porter's song...
 

Rodney Gold

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Dirac does not advise single measurement in any scenario. Even the armchair position requires 9. Single measurement effectively deprives Dirac of its mixed-phase feature, their unique selling point.
Trinnov measures differently thanks to the special microphone.
Yes, I know , but its sometimes a pain in the ass to do 9 sweeps , especially if you messing with speaker/ chair position etc ..
 

oivavoi

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My Swarm XO is usually at 100Hz in mono. No problems whatsoever. But my mains run full range. I tried them with high pass filter too and that works fine, but I prefer not to put
any additional processing in the mains signal path. My speakers' midwoofer actually has no crossover whatsoever. Experiment, experiment, experiment as in Cole Porter's song...

Nice. What speakers are you using, if I may ask?

My coming setup will anyway use a digital crossover for the mains, so no prob crossing over to subs in mono. Have you experimented with different slopes for crossing over also? 48 vs 24 db for example?
 
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pirad

pirad

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I had one midway all walls ... didnt stack them , ie mid front , back , side1 , side 2 .. my main speakers have 2 side firing woofers each .. so essentially 8 bass sources
Im pretty sure he had setup instructions somewhere or it might have been in a review or write up. The site is changed since I looked at it last

I dont use a swarm now ..albeit it did what it said on the tin , the subjective bass was not as good as the speakers alone and the upgrade to G1 spirits produces even better bass

So you are using one of Harman's Todd Welti configurations. See this paper page 76.
multsubs_0.pdf
I am curious how you take care of the phase of all those 8 drivers? Welti was using monopole sources, less possibility of unwanted suckouts.
I personally follow Earl Geddes research as implemented by his student and co-worker Duke LeJeune. See here:
http://www.gedlee.com/Papers/multiple subs.pdf
In 3 pages he summed up his PhD thesis he did on subs. In a nutshell:
one sub in the corner, one in the vicinity of the mains, the other 1-2 doesn't really matter as long as it's random.
There is math statistics to justify that, but perhaps one needs a PhD to grasp it. I love the simplicity of the solution,
and look/hear -- no EQ!
G1 Spirits are great speakers.
 

Rodney Gold

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So you are using one of Harman's Todd Welti configurations. See this paper page 76.
multsubs_0.pdf
I am curious how you take care of the phase of all those 8 drivers? Welti was using monopole sources, less possibility of unwanted suckouts.
I personally follow Earl Geddes research as implemented by his student and co-worker Duke LeJeune. See here:
http://www.gedlee.com/Papers/multiple subs.pdf
In 3 pages he summed up his PhD thesis he did on subs. In a nutshell:
one sub in the corner, one in the vicinity of the mains, the other 1-2 doesn't really matter as long as it's random.
There is math statistics to justify that, but perhaps one needs a PhD to grasp it. I love the simplicity of the solution,
and look/hear -- no EQ!
G1 Spirits are great speakers.

I did mess around with placement , all in corners , random , adjusting sub height etc and found the midwall position to sound best. I never really bothered with steep suckouts as I found them relatively inaudible.
I tuned the whole thing by ear ultimately after meauring but there are tons of settings and things you can change which led to never ending "fidellitis and tweaking" .. the trinnov is WORSE in that department as it gives you many many options in its correction and they all sound different
My biggest problem is that you don't know what is "right" as you have no real reference as to what the recording should sound like ... so despite all the measurements and maths , subjectivity is the final arbiter
 

Wombat

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I like plug-and-play. This is all-too-complicated and restricted to a small(domestic) but not-in-agreement minority. Lots of column inches looking for a practical, or otherwise, conclusion for the few. Is it really relevant in the general personal audio context?

Lots of words and not much offering for the general public. Now, audio auditorium design is a more pertinent subject.
 

Purité Audio

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Having imported Trinnov in their early years, tried Dirac, ( via MiniDSP and Amarra) , Genelec’s GLM, played round with Sonrworks and Acourate.
Is that less is definitely more, I prefer to deal with just really major issues , speakers such as the Kiis and 8Cs sound better to me than trying to correct traditional speakers after the fact.
MQA is just tosh.
Keith
 
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pirad

pirad

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Nice. What speakers are you using, if I may ask?

My coming setup will anyway use a digital crossover for the mains, so no prob crossing over to subs in mono. Have you experimented with different slopes for crossing over also? 48 vs 24 db for example?
I use my own construction dipole main speakers.
Slopes--experiment! The LPF needs to cut out the lows quickly, eg. Linkwitz-Riley 24dB is good.
The HPF for mains can be assymetric, find whatever merges them best.
 

Burning Sounds

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I use my own construction dipole main speakers.

In your first post you mentioned Linkwitz LX521 speakers - is this what you are referring to or have you designed your own dipoles?

I also wondered when you were experimenting using a Mytek MQA DAC (2 channel) were you then outputting an analogue signal to the MiniDSP 4x10 to get the 8 channels for the LX521s?
 
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pirad

pirad

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Oh, no? I am afraid that I must disagree based on my own experience. Mike position affects measured acoustic response. Measured individual acoustic responses affect the spatial average. So, if mike positioning is not the same before and after calibration, I have no idea what in the hell you have got in the way of a comparison. The after measurement is quite pointless unless the mike positions duplicate the before positions fairly exactly.

No need for head in a vice thanks to spatial averaging. That is its purpose.

Sorry, but your casual, arm's length, hand held mike technique might very well make clear why you did not get satisfactory results with Dirac Live. It would not work with anything else, either. It seems you do not understand the technology and proper measurement. So, how can you expect the miracle of a misapplied, misunderstood tool making some positive impact on your sound?
Of course position of the pick-up device , ear or mic, affects the measured response. Move the mic slightly and you are getting a different sound. But why you don't register it so easily with you ears when moving the head? If one inch makes a difference, why Dirac does not give the exact positions for the mic (and the head)? What happens when you take three rounds of 9 measurements moving
the mic slightly, as it happens when the positions are not firmy fixed for repeats ? Which calibration is right? How many people can sit on the sofa when you do the "sofa" calibration? Is the sound
the same for all 3-4 listeners? How about the auditorium calibration?
Read the papers by Johannson and Brannmark at Dirac site. ("Further reading").
https://www.dirac.com/live-home-professional-audio-info
Note that even the number of measurements is not fixed at 9, instead of "equal" sign they use the wave for "approx".
Thankfully Dirac does not attempt to correct for late reflections. It's arguably the best DRC because it knows the limitations of the technology.
I don't use it because it adds no value in the acoustically treated rooms where my system usually plays.
 
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pirad

pirad

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In your first post you mentioned Linkwitz LX521 speakers - is this what you are referring to or have you designed your own dipoles?

I also wondered when you were experimenting using a Mytek MQA DAC (2 channel) were you then outputting an analogue signal to the MiniDSP 4x10 to get the 8 channels for the LX521s?
LX521 were the first speakers I built. Then over several years I developed the dipole concept into my own construction. I did not stop there and went on with an integrated amplifier to connect all pieces.
When I was experimenting with MQA I was already using my own speakers. They resolve better than LX521. The weak point of LX521 is the MiniDsp 4x10HD. What's the point of testing MQA decoding DXD 352kHz studio codec and then feed it to MiniDsp mediocre 96khz DSP? Also the output stage is not so good, surprisingly RCA SE better than the balanced XLR.
But frankly Mytek lost out in my tests to Schiit Multibit. Even the lowly Modi was better in my auditory tests than any DAC based on delta/sigma Sabre chip. For home listening my favourite is Gungnir.
When I feel really naughty and want to prove that most pop/rock music is poorly produced I employ Yggdrasil. The nice feature of Mytek is it's levels display. So easy to show with it that pop/rock programme signal constantly goes into clipping.
 

Fitzcaraldo215

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I've always felt this way, too, but I am beginning to look at it another way. If the purpose of the multiple spaced measurement mic placements is to supply enough information to create a relatively uniform response in the volume of space that they encompass. Now, if one measures anywhere in that space with any number of positions, the result will probably not replicate the "predicted" result but it should show a significant amelioration when A/B-ed with EQ bypassed. If it doesn't show that, the process failed.
Probably so, Kal.

You know well what I am about to say, of course, probably much better than I. But for others, as I see it, a mike measurement for each speaker at any position will include, inseparably combined, both local frequency response unique to that one mike position plus broader, general room response, which is also part of the measurement results for that speaker channel at other mike positions. Local response peaks/dips in the measurement for one speaker at one mike position are not necessarily smaller than more general ones. The difference is the local ones don't show up for the same speaker at all the multiple mike points. The averaging of response for multiple mike points will therefore be weighted toward the general response issues for that speaker channel in the room and not the individual local ones. The EQ filter determining EQed response for that speaker is then calculated using the average, which has deemphasized the local response issues, though not entirely eliminating them mathematically. But, this should be the aim of the EQ tool, and it should minimize the influence of variable local response based on a single point. Some say no, and prefer EQ tools based on single point measurement. But, I remain more convinced by the multi-point, spatially averaged approach. However, the multi-point, spatially averaged approach creates difficulties and potential inconsistencies between pre-and post calibration measurements.

I am not at all sure how much more uniform in terms of variability from point to point post-calibration response is compared to pre-calibration. Likely so, particularly in the deep bass by reducing room modal issues. But, maybe not, even if on average the response is improved by the EQ process. Some local response issues and their variability may persist as frequency rises.

Smoothing of the frequency response used as input to the filter calculation should also play a role in all this, ideally with less 1/X octave smoothing at lower frequencies and more smoothing at higher frequencies, where chaotic, reflective, narrow band comb filtering becomes more severe. AFAIK, all EQ tools use some frequency smoothing in generating their filters, but I do not know who does use this ideal variable smoothing as a function of frequency. However, most EQ tools, like Dirac, only provide a graph of results with equal smoothing up and down the frequency spectrum, maybe 1/6 octave or so. That may not be sufficient to see issues in the lower frequencies. Amir indicated awhile back that the JBL/Harman EQ tool does provide a response graph with less LF smoothing and more HF smoothing. So, presumably that tool uses that variable smoothing in calculating its filters.

Dirac's specific algorithm for spatial averaging and smoothing is proprietary, and it might possibly include more sophistication. I recall Audyssey touting their "fuzzy logic" method for spatial averaging. It was claimed, though details were hazy, to provide a better picture of the general response for each speaker channel, with further deemphasis of the local response issues in the filter calculation. They also provided no direct method of post-calibration measurement. Nor, does Anthem ARC.

Yes, I would expect the result of a followup measurement test tone sweep to validate some improvement in post calibration response, even if the mike points are inches off of what they were during the calibration. The specific mix of general resposone issues to local ones might well be somewhat different, however, particularly approaching the HF comb filtering region. But, is this more accurate than the predicted response graph? Unless the prediction is a manipulated or overly optimistic fraud - tough to prove - it may have some value as well. But, which is right: the prediction based on the specific measurements feeding the filter calculation, or, if it were available, a followup, similarly smoothed and averaged multi-point measurement but with potential mike position differences, or a single point REW measurement via a different measurement protocol, or none of the above? Are apparent differences between them significant, given they are based on different measurements?

I personally don't want to spin my wheels trying to reconcile or validate different measurements in the messy world of non-anechoic room acoustics. I agree about standing back, letting the EQ tool properly do its measurement thing, and generating its algorithmic idea of improved response. Then, I will subjectively judge the before and after response by listening while easily switching back and forth. Subject to all the caveats about expectation bias, etc., I reach a conclusion, often achieved somewhat more reliably by blind testing. For me, the sonic differences have been sufficiently obvious and beneficial that blind testing yielded no surprises, disappointment or undue agita. And, I found this much more valuable and convincing than comparing before/after graphs, though some useful things like difficult to correct deep bass nulls have been revealed in the predicted response curves.

For over 10 years in different rooms with different systems and with different EQ tools, including those of friends, I am quite sure I have achieved better sound in my room this way, as are my friends in their rooms. I am also convinced that diligent care for and understanding of simple, correct mike measurement techniques during calibration have improved on the calibrated result. Comparing different calculated filter sets in Dirac is a piece of cake. But, YMMV.

Yes, it is clear that the most significant improvements are in the deep bass, both theoretically and subjectively. So, I have no issue with those who prefer to apply EQ only at low frequencies.
 
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mitchco

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Yes, I've done the same with Acourate, but by your "vice versa" method - exporting the simulation into REW and then comparing it to the REW measurement - they are as you say, virtually identical.

It's also easy to drop the filter you have created into Acourate and re-measure and compare that to the simulation.

@mitchco if you don't mind answering a question here - is it possible to use Acourate for correction below Schroeder frequency only? I don't see a way to do it, but perhaps it can be done using the manual controls rather than the macros?

Yes, Acourate does partial correction: https://www.computeraudiophile.com/...re-walkthrough/?do=findComment&comment=232841
I am updating my book on a variety of DSP topics, including this one, but if you need the procedure sooner than later, Uli is very responsive on the Acourate forum.

Audiolense also does user configurable partial correction, being able to adjust the frequency and time domains independently. Audiolense also has a user configurable mixed phase target design, so this is not unique to Dirac...
 

Dilliw

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Thank you for your insights into this topic! I myself have been playing with an FDA (SMSL AD18), but it adds another DSP step into the chain and my subjective review has been that the results vary based on source material. With my analog pre-amplifier and class A amp the stages after the DAC are consistent for better or worse.

I don't think we are far away from a device that can take multiple inputs and file formats, process them one time for room correction, EQ, filtering, and whatever else you want, then skip the DAC and output via i2s to PWM amplification tailored for various drivers (high, mid, bass) . I'm sure some of the DIY guys have already done with a Minidsp card or something but I'm too lazy for that route. I'm going wait some more and go back to living with what is antiquated analog volume attenuation, amplification, and passive speaker crossovers because with those I know what I'm getting.
 
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