Well duh, isn't that what room correction systems are SUPPOSED to do?!?! However nothing in this world is perfect. And the idea of this thread is not to focus on the lowest frequencies. (Nor to answer by saying "use active crossovers or DSP**)
*I have a bucket list dream to make some high sensitivity PA-style woofer + big horn towers. I would like constant directivity, which then means the response drooping at ah isn't it -6 dB per octave IIRC?
- My big question really is can Audyssey/Dirac/ARC fix the drooping response of a constant directivity horn?*
- Baffle step compensation, instead of bothering about this (which is then bothered by room boundary proximities anyway) can room correction "fix" it?
- If the correction system is allegedly time-based, can it correct time misalignment between drivers?
*I have a bucket list dream to make some high sensitivity PA-style woofer + big horn towers. I would like constant directivity, which then means the response drooping at ah isn't it -6 dB per octave IIRC?
- I did ask this question to Audyssey but the answer was equivocal and not specific.
- The amount of boost/cut needed for a big horn could add up considerably, no?
Really the speaker should be designed to avoid problems with the room because once done, you cannot undo them. You seem to be taking the word 'correction' too literally here, and I wouldn't blame you.. it is a common occurrence, it's more of a marketing term.
You can fix a drooping response if it's the speaker itself that is in need of tonal adjustment.
You can fix a drooping response if it's the speaker itself that is in need of tonal adjustment.
The problem is that the room correction system cannot tell whether the sound it is measuring is direct sound or a combination of direct / reflected sound. Speakers with different power responses will deliver different mixes of direct / reflected energy, so the room EQ will not know how much EQ to apply to sound 'natural' (a speaker with a cd / horn combo and speaker with a dome tweeter will sound different at listening position if they are both EQ'd flat at listening position, due to the different ratio of direct to reflected sound)
The problem I have with the premise of these debates is they assume we all have the same standard, perfect hearing. At the end of the day, when it comes to personal sound systems it's all about 'the sound we like'. I've just given away my JBL Control Ones because I don't like them. My cheap Sonys offer more punch, who knew?
Many systems can within reason, but keep in mind that latency can add up when you go to extremes with DSP. You often see video-specific, low-latency options in menus of commercial products doing this kind of correction.
As discussed earlier, many people view DSP correction as something that should be used judiciously. The more extreme the correction, the more likely there will be audible side effects. So it's often seen as desirable to do as much other stuff right as you can, so you aren't trying to dig yourself out of a big hole with DSP. It's a good idea to try to use corrections that are valid for multiple axes also. Single-point optimizations can be unnatural.
To have a better chance for correcting speaker problems I'd opt for programs like DRC-FIR/Audiolense/Accourate and not the products mentioned in the first post.
The 3 products I mention all have user adjustable options to get closer to actual speaker correction instead of "Room correction". Room correction is a nice term, but it shouldn't be used for products like these. They can help make the combination of speaker + room behave better, but it is the job of the man behind the tool to know and/or learn what to correct and what not to correct.
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The tools mentioned in the first post are rather 'closed box' solutions that don't offer much user input, I'd say that leads to plug and pray.
DSP often gets a bad reputation. Unneeded if you ask me if you spend the time learning what it can, and what it should not correct. If used well with a good speaker design it can work wonders. But the key is to start with something that has a shot of being good. Solve everything you can with passive means first, put DSP on top for the icing on the cake.
DSP cannot correct a bad speaker design. It cannot alter the DI of a speaker either. It can do most of what you mentioned in the first post, but make sure you know what you want it to do. That's a way bigger challenge.
The 3 products I mention all have user adjustable options to get closer to actual speaker correction instead of "Room correction". Room correction is a nice term, but it shouldn't be used for products like these. They can help make the combination of speaker + room behave better, but it is the job of the man behind the tool to know and/or learn what to correct and what not to correct.
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The tools mentioned in the first post are rather 'closed box' solutions that don't offer much user input, I'd say that leads to plug and pray.
DSP often gets a bad reputation. Unneeded if you ask me if you spend the time learning what it can, and what it should not correct. If used well with a good speaker design it can work wonders. But the key is to start with something that has a shot of being good. Solve everything you can with passive means first, put DSP on top for the icing on the cake.
DSP cannot correct a bad speaker design. It cannot alter the DI of a speaker either. It can do most of what you mentioned in the first post, but make sure you know what you want it to do. That's a way bigger challenge.
I haven't tried to use Audyssey or Dirac to do what a good DSP crossover can do with REW and a UMIK-1: EQ a controlled directivity horn/compression driver to flat response in my listening room.
There is good reason for my lack of trust: these two firmware/software products cannot do a good job in my listening room with full-range horns having directivity control down to 100-120 Hz--a setup that's already been dialed-in using DSP crossovers (using REW, etc.). I have tried many times over the years to get these firmware/software products to produce even a listenable result and have failed. I can give you an example of what even Dirac Live Full does to my setup (I captured the left front loudspeaker a 5.2 fully horn loaded array in a relatively live room--RT30 response < 0.5 s across the audible frequency spectrum).
So if you've gotten any of the three products you mentioned to work at all with cone/dome loudspeakers having poor directivity control, consider yourself extremely lucky. I haven't had any luck with fully horn-loaded loudspeakers--a task that should be much easier for the software/firmware products to work well. The frequencies that were adversely affected were from 50 Hz to over 1000 Hz.
I would take wesayso's recommendations to heart. Anything that measures further away than 1m for each loudspeaker in-room (unless you own an extremely dead listening room with RT30<0.1 s), and temporarily using plenty of absorbent material on the floor between the microphone and the loudspeaker, you're not measuring the minimum phase performance of the loudspeakers--and these "room correction software" packages can't seem to pick out the minimum phase behavior that it can correct from the mixed phase/all pass behavior in-room. You can see this in the resulting phase response of the measurements, which will wrap up from -180 to 180 degrees many times even across one horn/driver combination if trying to measure at the listening position(s).
Chris
This is a good point. Supposedly Audyssey (and the others I'd guess) can measure the direct sound, but if you are at a long distance this will only work at the highest frequencies. Conversely I guess if you are listening that far back, the speaker and room sound are merged together anyway.
For me, the problem with DSP crossovers or the programs that @wesayso mentions is those are not turnkey compatible with multichannel at this house. Due to space and time I would never get around to implementing a home theater PC (and I'm much more a Mac guy anyway). I wish some intrepid company would make a digital out decoding processor...not likely given the investment needed 🙁 hence my wondering if "room correction" in an AVR would correct the response droop out of a CD horn.
I'm not sure that you've considered the following:
1) Using an AV preamplifier/processor and DSP crossovers, with separate amplifiers, like the following connection diagram:
(More on this subject here.)
2) Then use Room EQ Wizard to measure and provide semi-automatic inputs to your DSP crossovers, like the following:
Using REW to Find Parametric Equalizer (PEQ) settings
[The reason for the "semi-automatic" nature of using REW is due to the issues trying to make it "fully automatic" and coming out with something that sounds terrible.]
3) To get the proper time delays on the individual driver channels within a loudspeaker, you can use this method:
Using REW to Determine Time Delays Between Drivers
4) You can use the "Room Correction Software" packages in your pre/pro to set the multichannel array (5.1, etc.) loudspeaker channel gains and time delays.
Voila! Now you have essentially full control of the resulting sound and full visibility of what it's doing as well as the ability to tweak the DSP crossover or pre/pro channels individually to achieve your goal.
Chris
I think it is a simple fact - room correction is not something for the lazy.
I have followed Chris A's recommendations measuring close to the loudspeaker. My system is a stereo pair of Chris A's MEH which exists by the good graces of Tom Danley's and Roy Delgado's great foundational work. Chris is very modest about his part in this but I am enamored of what he did and the great amount of time he put into it notwithstanding his tutorials on how to make one for oneself.
Measuring at various distances shows you how things change with distance - you see how phase broadens with distance from the speaker.
I start with measurements at 1 meter from the beginning of the horn (K402). I find the tweeter adjustments to hold as the distance is increased - not perfectly by any means but I have found it best to leave well enough alone.
Below 300 Hz one can see the room making its greatest effect. I make the each speaker as flat as possible which always results in the flattest phase plot as the first step.
Measuring from my chair, in the traditional equilateral position from the loudspeakers, there will be peaks that can be tamed and this is clearly audible. These peaks can be surprisingly high in amplitude and high Q. I no longer try to flatten the response from this position even though I can using the xilica SOLARA - one can have as many PEQs as one wants - though one hears this is not the way to go. I think there is advantage to having as many PEQs as possible just to see for oneself the damage this can do. These adjustments are made with both speakers playing. My system is a simple two channel setup whereas Chris A's system is multiple channels. I tend to think, not from my own experience, that multiple channels is good for the sound above the bass just as we know that multiple subwoofers allow for a more even bass response. I use four subwoofers, one in each corner of the room. I suspect that a multi channel setup is able to tame the room problems better than two channel. I cannot afford more channels and have no place to put the boxes. All of us have to do the best we can with what we have.
All of this requires literally thousands of sweeps. One must go back and forth between "hearing" what the adjustments display and actually listening to the effects. The microphone does not hear like the ear does. One has to learn to interpolate and extrapolate.
All of the automatic devices might be better than nothing but not much more than that. If it sounds too good to be true it most likely is. I have never tried one of these - I am too cheap to buy them since I think the result will not be satisfactory. Plus they require having a full fledged computer in the signal path. I have used both the silica XP series and the SOLARA and whatever toll these devices take is more than made up for with clearly audible improvements. They are quiet unlike some of the cheaper devices. I do use a very simple computer as my digital source.
I have found that for measurements below 200 Hertz all speakers must be playing that are involved in this region and adjustments are made for all of the speakers that are exactly the same. I have tried numerous times trying to adjust the woofers individually (in my case RYTHMIK 15s) and the reuslt is never anywhere as good as one of the rare cases where one size fits all. Of course each speaker is loading the room in a different way individually but all seems to average out with all of them putting the same energy into the room. Of course, my experience is not going to be the same for everyone.
I do include the bass channel of the MEH in these measurements and adjustments - they are in addition to the baseline 1 meter adjustments made initially.
Learning how to use the magnificent tool that is REW is fun but all of this is not for those without the time or inclination to stick with it and put in the hours and hours to get a good result.
I have followed Chris A's recommendations measuring close to the loudspeaker. My system is a stereo pair of Chris A's MEH which exists by the good graces of Tom Danley's and Roy Delgado's great foundational work. Chris is very modest about his part in this but I am enamored of what he did and the great amount of time he put into it notwithstanding his tutorials on how to make one for oneself.
Measuring at various distances shows you how things change with distance - you see how phase broadens with distance from the speaker.
I start with measurements at 1 meter from the beginning of the horn (K402). I find the tweeter adjustments to hold as the distance is increased - not perfectly by any means but I have found it best to leave well enough alone.
Below 300 Hz one can see the room making its greatest effect. I make the each speaker as flat as possible which always results in the flattest phase plot as the first step.
Measuring from my chair, in the traditional equilateral position from the loudspeakers, there will be peaks that can be tamed and this is clearly audible. These peaks can be surprisingly high in amplitude and high Q. I no longer try to flatten the response from this position even though I can using the xilica SOLARA - one can have as many PEQs as one wants - though one hears this is not the way to go. I think there is advantage to having as many PEQs as possible just to see for oneself the damage this can do. These adjustments are made with both speakers playing. My system is a simple two channel setup whereas Chris A's system is multiple channels. I tend to think, not from my own experience, that multiple channels is good for the sound above the bass just as we know that multiple subwoofers allow for a more even bass response. I use four subwoofers, one in each corner of the room. I suspect that a multi channel setup is able to tame the room problems better than two channel. I cannot afford more channels and have no place to put the boxes. All of us have to do the best we can with what we have.
All of this requires literally thousands of sweeps. One must go back and forth between "hearing" what the adjustments display and actually listening to the effects. The microphone does not hear like the ear does. One has to learn to interpolate and extrapolate.
All of the automatic devices might be better than nothing but not much more than that. If it sounds too good to be true it most likely is. I have never tried one of these - I am too cheap to buy them since I think the result will not be satisfactory. Plus they require having a full fledged computer in the signal path. I have used both the silica XP series and the SOLARA and whatever toll these devices take is more than made up for with clearly audible improvements. They are quiet unlike some of the cheaper devices. I do use a very simple computer as my digital source.
I have found that for measurements below 200 Hertz all speakers must be playing that are involved in this region and adjustments are made for all of the speakers that are exactly the same. I have tried numerous times trying to adjust the woofers individually (in my case RYTHMIK 15s) and the reuslt is never anywhere as good as one of the rare cases where one size fits all. Of course each speaker is loading the room in a different way individually but all seems to average out with all of them putting the same energy into the room. Of course, my experience is not going to be the same for everyone.
I do include the bass channel of the MEH in these measurements and adjustments - they are in addition to the baseline 1 meter adjustments made initially.
Learning how to use the magnificent tool that is REW is fun but all of this is not for those without the time or inclination to stick with it and put in the hours and hours to get a good result.
You can't fix a speaker with EQ alone. It can help a good speaker eke out that nth percentile, it can fix phase, it can help a response, but you always butt into the limitations of the speaker itself. EQ can't make the magic triumvirate of small, loud and clean-- You still have to pick two. I use EQ to reduce room rattles more than anything. Does my lamp reverb at 225hz? I have a narrow PEQ to reduce that. My door rattles at 50hz? Another one. A better room would be nicer, but EQ is free, making it tempting to anyone.
I'd agree with that in general. What do you think about EQ to counteract the falling high frequency response of a constant directivity horn?
I'd considered it, with the "evil" that there is another A/D step usually at a low volume input to the DSP crossover. I don't think that is automatically the end of the world but it doesn't seem desirable. I wish somebody would make a digital out pre/pro but Apple will be selling us direct neural interface via Nikola Tesla energy conduction from Apple "BrainPods" before that happens.
I looked at the Home Theater PC forum and that was the end of that, OMG I'm not putting that kind of time into it, too many other things going on.
Another alternative would be preamp out into analog crossovers...perhaps that's still better than passive with big woofers and horns. Then again it seems silly to have an AVR with like 100 watts, more than enough to drive a sensitive speaker to high volume, and then throw that away to go buy external amps. Ah well nothing unusual is easy.
The need for a AVP is to be able to handle the different A/V codecs to decode 5.1/7.1 etc. channels, and the legal requirement to have HDMI for carrying DSD from any SACDs (stereo or multichannel) you might own--to avoid a conversion of DSD to PCM in a source player which actually is audible in my experience.
If you're not intending to use those types of sources and multichannel operation, then a simple digital output source (your choice) into DSP crossovers (using S/PDIF or AES3 inputs) eliminates the A/D-D/A (which I've found to be transparent/inaudible in any case). I use a good external two-channel DAC for two-channel-only operation with HDMI switches for bypassing the AVP. That could easily be replaced by a source with S/PDIF into DSP crossovers with digital inputs.
There are also ways to go from PC directly to DSP crossovers via AES3 output card, if you've got the multichannel issue under control in the source PC. I've got an acquaintance that has been doing that for a number of years with an RME HDSPe AES card.
Chris
Sure, but you always sacrifice headroom somewhere else. You're sacrificing the vaunted sensitivity of compression horns for an optimal on-axis and potentially nice off-axis if DI is still correct all the way up there.
I didn't deal with Audyssey/ARC, so Dirac as example.
Dirac this is first of all EQ, of course with EQ you can correct CD response, the easiest part, just add HF.
This is a bit more difficult... But in general, this is also corrected by EQ, so why not.
Correctly performed measurements in Dirac can help. There is an array of 9 dimensions available, if I remember correctly, it is important to perform them from random spots so that the averaged curves are less dependent on the interaction with the room. If you take measurements too close to one point in space, you will accumulate room errors of the same type. Comparison with measurements in the REW program helps to better understand what is happening.
Dirac is able to more or less correct the group delay of the MF/HF crossover as FIR filter. But of course, it can't magically transform a bad crossover into a good one. And you cannot control it, only check the result.
Generally, the tasks you described can be more or less solved or at least improved. Another question, it is not a fact that this will happen automatically.
And of course, you can not pay money for this software, but use REW, Rephase, some kind of DSP / plugins. And REW control will help to squeeze all the juice out of any software tool, you will not act blindly.
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I'm not sure what you're thinking here...but in my experience, this is a red herring. For instance:
Neither of the above raw response curves on a K-402 horn have any effect on the dynamic range or subjective sound quality of the flat-EQed loudspeaker performance. In fact, the red curve (raw response before EQ applied) is exactly what Klipsch now uses in its flagship product--the "Heritage" Jubilee ($35K+ USD per pair), crossed at ~250 Hz [second order L-R] to the bass bin and extending to 20 kHz:
An on-axis acoustic transfer function plot (no smoothing) of that compression driver/horn combination (in-room measurement at 1 m) follows:
A polar sonogram plot of the K-402-MEH using another 2" compression driver (K-69-A) below. The green-to-blue color transition is the now-standard -6 dB directivity coverage angle value used in EASE, etc.:
Chris
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Dirac etc. are used to correct for farfield response and room echoes/modes' spl wiggles. Like said earlier, they cannot fix a bad loudspeaker design.
With multiway speakers you need multichannel dsp to integrate units' responses and timing. Directivity comes from driver and xo-freq choices.
When making gross corrections with dsp ome must be careful not to boost too much, that will cause digital clipping. Eg. horn's low freq must be attenuated.
Bass driver's capacity down low (Xmax or BR tuning) can also become the limiting factor.
Multichannel dsp is tricky, you must understand general rules of loudspeaker design, limits of dsp and add difficulties when taking measurements.
ADC and DAC back and forth is not problem per se with modern chips, but poor settings are! Like Cask05 I use a multichannel AVR as source/preamp, diy multichannel-dsp active speakers are just for main LR (stereo)
With multiway speakers you need multichannel dsp to integrate units' responses and timing. Directivity comes from driver and xo-freq choices.
When making gross corrections with dsp ome must be careful not to boost too much, that will cause digital clipping. Eg. horn's low freq must be attenuated.
Bass driver's capacity down low (Xmax or BR tuning) can also become the limiting factor.
Multichannel dsp is tricky, you must understand general rules of loudspeaker design, limits of dsp and add difficulties when taking measurements.
ADC and DAC back and forth is not problem per se with modern chips, but poor settings are! Like Cask05 I use a multichannel AVR as source/preamp, diy multichannel-dsp active speakers are just for main LR (stereo)
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