You just said it. ...Not only acoustic room treatments are required in addition to parametric equalizers, but also microphones (@ various room's positions - listening area) and DSP equalization with FIR and IIR digital filters.
All the room's reflections, reverberations, all the crossovers between the drivers, all the various frequency responses (bass - mid - treble) cannot be fixed by parametric equalizers alone. ...Just too many acoustical delays (phase all over the place). ...Phase of everything; the drivers in the speakers, the electronic x-overs, walls, ceiling, floor, furniture, various materials, the speaker's locations, the subwoofer's locations, the listener's position, the recordings, the source playback machines, the electronic circuit boards inside them machines, the implementation, the wiring, the cables used, the power cords, everything!
We know so very little still about proper sound propagation, reproduction, and propulsion. ...Before we equalize we should know first the acoustic properties.
...And how they are perceived in time and space for a dedicated room's size of a certain composition and correlation with someone's hearing.
And the golden rule of averaging is as good as the ears we accept to use as the ultimate guide. ...But still only a guide, and never absolute.
The standards we create are always to be explored further, and not the scientific audio bible. In sound as in picture we are way behind the reality.
Are those parametric equalizers?
Did I not also mention measurement tools, and the pursuit of always improving them?
Plus how they correlate (accurate sound measurements) with our perfect hearing.
Yes you did, but I don't think you actually had any equipment in mind, and I don't know why you're bringing up correlation with hearing again. It was suggested that we should look extremely closely at phase measurements over a few degrees to see that speaker drivers are not minimum phase, I replied asking how we are supposed to take a measurement accurately enough to do that. Simple suggestion, simple question, and based on your two responses, I'm pretty sure you have nothing to say about either. You don't need to reply to every post in this thread, you know.
1. I will bring whatever I consider important to my ears and mind.
And no I didn't have any specific piece of electronic in mind.
2. It cuts both ways. I reply when I got something to say that I consider worthy of saying.
3. I'm not an audio scientist, I'm not a sound engineer, I'm not an acoustician; I'm just me, a sound explorer in search of answers.
* Watch the Winter Olympics Games; it's relaxing for the soul. 🙂
I let Markus, or Mitch, or any other member reading this and interested to give an answer or two, to reply.
I thought, I tried, I suggested, just to advance the discussion by having other members interested in their own reactions, suggestions, ...
Sean Olive's blog said:
Since we know the Harman system uses PEQ to do the Amplitude correction, how do they incorporate the "Time" correction part?
Would it be something like using minimum phase PEQ followed by some sort of phase correction?
For a while I thought about using a linear phase PEQ to correct Amplitude while preserving Time, but would it be a smarter choice to use minimum phase PEQ and correct phase afterwards? The linear phase PEQ might get me in to much trouble with pre-ringing?
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^
Minimum phase IS the "time correction part", IF the response is minimum phase to begin with.
Whether a certain frequency range is minimum phase (enough) can be seen in a excess phase group delay graph. See Minimum Phase
Linear phase is probably a bad idea because of filter pre ringing. Mixed phase could be used if pre ringing is kept below audible threshold and if the filter is spatially robust within the listening region. The latter is also true for minimum phase filters.
Minimum phase IS the "time correction part", IF the response is minimum phase to begin with.
Whether a certain frequency range is minimum phase (enough) can be seen in a excess phase group delay graph. See Minimum Phase
Linear phase is probably a bad idea because of filter pre ringing. Mixed phase could be used if pre ringing is kept below audible threshold and if the filter is spatially robust within the listening region. The latter is also true for minimum phase filters.
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You guys realize that phase has meaning only if one is using data that contains more than one or two reflections. It's basically random. The difference in MP versus non-MP filtering only has a meaning in the context of anechoic (or gated) based EQ. For steady state based data only MP (IIR) filters make sense.
To my ears, linear phase FIR filters sound best for stereo localization. Acourate uses preringing compensation (implemented 3 years ago) to keep preringing below one's audible threshold.
Markus, you have Acourate, why not make minphase and linphase correction filters and compare the two? If using JRiver's Convolver, one can easily AB the correction filters while listening to music, plus compare to PEQ. Report back which one sounds best to your ears.
Some theoretical discussion on this thread, but little discussion on people's actual listening experiences with modern digital room correction software (or PEQ) and filter types...
We're talking about steady state in-room responses. Every response within the listening area represents a linear system. If there are minimum phase regions that are the same for each and every system (look at the excess group delay) then all system responses can be improved with a single filter.
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I'm not interested in discussing what sounds good to a random poster on the internet. Listening tests are rather meaningless if biases aren't controlled.
How about some vintage analog gear?
A guy named Paul Kemble recommends SH-9010 - 5-band PEQ from the 70ies by Technics
A Paul Kemble web page - equalisation.
I have two of them and checked them with DEQ RTA function. They appear to work as advertised indeed - reliable and very precise.
A guy named Paul Kemble recommends SH-9010 - 5-band PEQ from the 70ies by Technics
A Paul Kemble web page - equalisation.
I have two of them and checked them with DEQ RTA function. They appear to work as advertised indeed - reliable and very precise.
I have set up several listening tests and reported both measurement and listening results here: http://www.computeraudiophile.com/blogs/mitchco/
Each test is designed to be repeatable so that anyone can set them up and listen for themselves and draw their own conclusions. For example, here is repeatable test to determine how far from bit-perfect one can detect an audible change in SQ playback: Computer Audiophile - Fun With Digital Audio – Bit Perfect Audibility Testing
My thoughts, including measurement and listening results are here: Advanced Acourate Digital XO Time Alignment Driver Linearization Walkthrough
If you are not interested in discussing measurement and listening results for room correction, and the correlation thereof, why bother posting your question then?
All the best, Mitch
Another experience FYI. My approach has been similar to what wesayso mentioned:
> Use DCX for active XO with delays and filters chosen to achieve phase agreement throughout the XO ranges (SWs-MW and MW-TW).
> Use DCX for PEQ (IIR's) as needed for direct signal EQ and a room mode issues. [Sometimes I also judiciously address a couple of the 100-600 Hz room effects.]
> Use RePhase to design an IR to remove the phase rotation of the direct signal without amplitude impact.
> Use Foobar to provide the convolution.
This method addresses the EQ "needs" and any "wants" without adding any preringing. The phase rotation is also removed without adding any preringing.
Some Charts:
Since my LP is at 4 m I pick up lots of room impact on the phase measurements. The 24" phase of FL and FR is shown below to better show the direct signal phase. When the TW and MR are measured very close (not shown) the phase ripple is nearly completely removed as the impact of the diffraction and reflections is very small.
jtalden, very nice! Subjectively, I bet your system sounds great ;-)
Neat use of RePhase. Note that using FIR filters the phase can be corrected independently of the amplitude: FIR vs IIR filtering | MiniDSP
Attached is the step response of my system using linear phase FIR filters. Note no preringing.
I will fire up REW and take a measure with the FIR filters in circuit and compare results. In Acourate, looks very similar to what you have in REW, but would be interesting to compare using the same measurement software.
Again kudos, very nice! Thanks for sharing another approach that obviously (to me anyways) works well. Cheers!
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...And jtalden for the good post above!
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