INTERACT FORUM
Devices => Sound Cards, DAC's, Receivers, Speakers, and Headphones => Topic started by: 6233638 on November 30, 2013, 08:11:39 am
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I was wondering if anyone here had advice or recommendations about room correction.
I don't have a particularly high-end speaker setup right now, as I have mostly been focused on headphones the last few years, but I'm now looking to move in that direction.
Before buying better speakers though (I have a pair of OK bookshelf speakers right now) I was wanting to set up some room correction first and see how good I could get those to sound. After all, higher-end speakers should benefit from room correction just as much as the ones I have now.
I've been experimenting with the demo of Audiolense as it lets you take measurements and apply the corrections to a 90-second music sample, and it's definitely making an improvement.
Right now my setup is all a bit thrown together though. I'm using my old Radioshack 33-4050 SPL meter with a generic calibration file hooked up to the line-in of my PC's on-board sound, and because I wasn't able to find any any long audio cables, I hooked up my Benchmark DAC2 through a long USB extension and used that as the source. (latency was too high via Airplay)
So I have some idea of what room correction is going to do for me, but I'm a bit unsure about how to proceed now.
For now at least, I'm only interested in stereo (2.0) correction, but I will probably want to correct the audio in multiple rooms.
I think this rules out Dirac as I believe they use a driver to process the sound locally on the PC, rather than offering a VST plugin or working with Media Center's Convolver.
Unfortunately, that was probably going to be my first choice as I've heard the Dirac software in use before, and it was very impressive.
As far as I know, both Acourate and Audiolense should work via Media Center's Convolver, and I'm not sure if there are other options that do as well.
At first, Audiolense appears to be the cheaper option at $225 for the 2.0 version, but it seems that the $530 XO version actually includes features which would still be relevant to a stereo setup?
On the other hand, there's only one version of Acourate and it's $390, which keeps things simple.
It seems like Acourate is the more advanced package, and comes highly recommended, but it's a lot of money when there are no functional demos for it, and Audiolense offers you three months to return it if you're unhappy with the results (http://www.juicehifi.com/index.html).
I assume that if I'm already spending that much on the software package, I should also be replacing my SPL meter with a calibrated mic - which seems to be in the $150-400 range.
I'm not quite sure what to do about sound hardware though. I've read that you should be using the same sound device for both playback and recording for time domain corrections to be performed correctly. Is that true? I would probably have to invest in more sound hardware if that's the case. (or use the on-board audio)
It also looks like Acourate requires you to use a single ASIO device (no MME/DirectSound support) for both input and output.
It seems like you could probably use something like ASIO4All to use any devices you like with it, but there's probably a reason it does things that way?
I was thinking of buying the XTZ Mic that Dirac recommend (http://www.xtzsound.us/shop-en/measurement-system/xtz-microphone-to-dirac-live-room-correction-suite) as it's only $140 and is a USB device, unlike most of the other calibration mics being sold - but that probably wouldn't work with Acourate (at least not without ASIO4All) and not if you need to be using the same sound device for playback and recording in these packages.
Another thing I was wondering about was room position. Both Dirac and Audiolense let you take multiple readings from different positions in the room so that it sounds good everywhere rather than only where the Mic was when taking measurements. Is this something Acourate offers as well? Is this feature all that necessary?
I know that I've only mentioned Acourate, Audiolense, and Dirac, but that's because it's all I'm aware of. I've not ruled out anything else if there's other software you would recommend instead.
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I assume that if I'm already spending that much on the software package, I should also be replacing my SPL meter with a calibrated mic - which seems to be in the $150-400 range.
I'd recommend getting a condenser microphone of some kind no matter what software you're using; even an uncalibrated mic will probably get you farther than an SPL meter (depending on the meter). This is what I use currently (it ships with a measurement software suite that provides some useful utilities): http://www.parts-express.com/dayton-audio-omnimic-v2-precision-measurement-system--390-792. I used phantom-powered mics for a few years, and after having a USB mic around for a while I'm never switching back if I can help it.
I'm not quite sure what to do about sound hardware though. I've read that you should be using the same sound device for both playback and recording for time domain corrections to be performed correctly. Is that true? I would probably have to invest in more sound hardware if that's the case. (or use the on-board audio)
It also looks like Acourate requires you to use a single ASIO device (no MME/DirectSound support) for both input and output.
It seems like you could probably use something like ASIO4All to use any devices you like with it, but there's probably a reason it does things that way?
Several of the automated correction suites make a similar recommendation. The reasons I've seen for using the same device for input and output relates to latency matching, and if the software doesn't support multiple interfaces, I wouldn't recommend trying to use ASIO4All to trick it into working (ASIO4All's latency is not predictable in my experience). That said, it seems like the basic problem would be a trivially easy problem to solve when running logsweeps or other artificial calibration tones. The software knows what the test tone is "supposed" to look like, how long it's supposed to last, etc. It seems like it would be simple enough to just look for a loose match in the input, and some measurement software cheerfully does this already. For example, Holm Impulse (which is freeware) doesn't care what your input and output devices are and still measures time-domain responses successfully, and it looks like Dirac has a way to do it as well.
Another thing I was wondering about was room position. Both Dirac and Audiolense let you take multiple readings from different positions in the room so that it sounds good everywhere rather than only where the Mic was when taking measurements. Is this something Acourate offers as well? Is this feature all that necessary?
The feature is pretty important for an automated system.
In the low frequency range there are a lot of issues that multiple measurements will help address, but a good example is Room Modes, which affect the perceived loudness of sound differently in different locations in the room: see http://en.wikipedia.org/wiki/Room_modes and http://en.wikipedia.org/wiki/Loudspeaker_measurement#Room_measurements. The nature of Room Modes is that different frequencies will seem to be amplified or nulled by different amounts in different places in the room. So if your measuring position is in a 200Hz null, an automated tool might try (possibly vainly) to fill that null. The result is that everywhere else in the room 200Hz will sound pretty loud, and the spectral balance will be goofed up. You can see how that would play out across different fact patterns. Taking into account multiple measurements will help to distinguish whether the frequency response non-linearity is a function of a specific position in the room, or whether it originates with your speakers (or with the listening area as a whole).
At high frequencies, the wavelength of sound becomes very small in relationship to us. For example, sound at 3KHz has a wavelength of about 12cm and 8KHz has a wavelength about 4cm. Performing phase correction on sound that localized based on one measurement means that even folks sitting right next to you may have a very different experience of the sound. Some folks argue that attempting to phase correct high-frequencies is a generically bad idea, but if you take multiple measurements you might discover a necessary common correction (i.e. a phase shift present most places in the room).
I know that I've only mentioned Acourate, Audiolense, and Dirac, but that's because it's all I'm aware of. I've not ruled out anything else if there's other software you would recommend instead.
I'd recommend you have a look at REW http://www.hometheatershack.com/roomeq/ It's not as powerful or flexible as the three you mentioned, but it has the advantage of being completely free, and might give you an opportunity to get your feet wet with measurement and room correction.
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I'd recommend getting a condenser microphone of some kind no matter what software you're using; even an uncalibrated mic will probably get you farther than an SPL meter (depending on the meter). This is what I use currently (it ships with a measurement software suite that provides some useful utilities): http://www.parts-express.com/dayton-audio-omnimic-v2-precision-measurement-system--390-792. I used phantom-powered mics for a few years, and after having a USB mic around for a while I'm never switching back if I can help it.
If you're in a pro environment and already own a Mic amplifier, it seems like a phantom powered Mic might be easier, but for this type of use, USB definitely seems the way to go as long as the device doesn't require specific drivers to work.
That looks pretty nice, and I like that it comes in a custom case - it's going on the list of mics I'm considering.
It's funny, I have spent so much on display calibration hardware, but for some reason I have difficulty spending a lot of money on audio calibration.
EDIT: It looks like the meter I have is a condenser mic. It's obviously not going to be on par with something like that of course.
Several of the automated correction suites make a similar recommendation. The reasons I've seen for using the same device for input and output relates to latency matching, and if the software doesn't support multiple interfaces, I wouldn't recommend trying to use ASIO4All to trick it into working (ASIO4All's latency is not predictable in my experience). That said, it seems like the basic problem would be a trivially easy problem to solve when running logsweeps or other artificial calibration tones. The software knows what the test tone is "supposed" to look like, how long it's supposed to last, etc. It seems like it would be simple enough to just look for a loose match in the input, and some measurement software cheerfully does this already. For example, Holm Impulse (which is freeware) doesn't care what your input and output devices are and still measures time-domain responses successfully.
You're right - I did some testing with my DAC hooked up to the line-in on the PC, and latency was jumping around from 40-60ms, to 300-400ms every few seconds when it was running through ASIO4All as a single device.
I agree that latency should not be affecting the measurements, but it definitely caused problems with some software when I was using Airplay.
With Acourate, you cannot use separate input/output devices, you can only select a single ASIO device and use its inputs/outputs. If Acourate is the software to get, I suppose I can pick up a reasonably cheap USB audio interface for those purposes.
I have been wanting to get a nice desktop mic at some point down the line, so perhaps something like a Focusrite Scarlett 2i2 would be a good purchase. (or complete overkill? I'm not sure)
The feature is pretty important for an automated system.
In the low frequency range there are a lot of issues that multiple measurements will help address, but a good example is Room Modes, which affect the perceived loudness of sound differently in different locations in the room: see http://en.wikipedia.org/wiki/Room_modes and http://en.wikipedia.org/wiki/Loudspeaker_measurement#Room_measurements. The nature of Room Modes is that different frequencies will seem to be amplified or nulled by different amounts in different places in the room. So if your measuring position is in a 200Hz null, an automated tool might try (possibly vainly) to fill that null. The result is that everywhere else in the room 200Hz will sound pretty loud, and the spectral balance will be goofed up. You can see how that would play out across different fact patterns. Taking into account multiple measurements will help to distinguish whether the frequency response non-linearity is a function of a specific position in the room, or whether it originates with your speakers (or with the listening area as a whole).
At high frequencies, the wavelength of sound becomes very small in relationship to us. For example, sound at 3KHz has a wavelength of about 12cm and 8KHz has a wavelength about 4cm. Performing phase correction on sound that localized based on one measurement means that even folks sitting right next to you may have a very different experience of the sound. Some folks argue that attempting to phase correct high-frequencies is a generically bad idea, but if you take multiple measurements you might discover a necessary common correction (i.e. a phase shift present most places in the room).
I'm not sure if Acourate can do this or not - I just noticed that Audiolense and Dirac made it clear that they did use this in their corrections.
I'd recommend you have a look at REW http://www.hometheatershack.com/roomeq/ It's not as powerful or flexible as the three you mentioned, but it has the advantage of being completely free, and might give you an opportunity to get your feet wet with measurement and room correction.
You know, I did give REW a shot last night, but I couldn't figure out how to create correction filters with it, though that was before I switched from using Airplay to my Benchmark DAC, so perhaps that was the problem. (latency was causing all sorts of problems in REW)
I've only ever used REW for general measurements and manually doing some EQ or adjusting the gain/crossover on a sub before.
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If you're in a pro environment and already own a Mic amplifier, it seems like a phantom powered Mic might be easier, but for this type of use, USB definitely seems the way to go as long as the device doesn't require specific drivers to work.
That looks pretty nice, and I like that it comes in a custom case - it's going on the list of mics I'm considering.
It's funny, I have spent so much on display calibration hardware, but for some reason I have difficulty spending a lot of money on audio calibration.
EDIT: It looks like the meter I have is a condenser mic. It's obviously not going to be on par with something like that of course.
Well that's good news. If you've already got a condenser mic, then you could put off the mic purchase until you've settled on a software package/audio interface.
You're right - I did some testing with my DAC hooked up to the line-in on the PC, and latency was jumping around from 40-60ms, to 300-400ms every few seconds when it was running through ASIO4All as a single device.
I agree that latency should not be affecting the measurements, but it definitely caused problems with some software when I was using Airplay.
Oh, don't get me wrong, I'm sure it is a problem for some software; I just noted that the problem should be solvable, and some software is insensitive to that issue or includes a solution for it. Software that expects you to use the same device will probably have problems if you use it with different devices.
With Acourate, you cannot use separate input/output devices, you can only select a single ASIO device and use its inputs/outputs. If Acourate is the software to get, I suppose I can pick up a reasonably cheap USB audio interface for those purposes.
I have been wanting to get a nice desktop mic at some point down the line, so perhaps something like a Focusrite Scarlett 2i2 would be a good purchase. (or complete overkill? I'm not sure)
I think the Scarlett would work fine for these purposes (I think Glynor has one of it's bigger cousins, and Matt just picked up a Scarlett 18i20), you could also use something like the Steinberg UR22. Before you get too far down that road, though, you had mentioned in another thread that you were thinking about bi-amping, but with a sub. If that's still on the table you should try and find an interface that has enough outputs for that project too. The focusrite 2i4 or 6i6 both have four analog outputs, the 18i20 has eight outputs. The Steinberg UR824 I mentioned in the other thread has 8 outputs as well. All of those interfaces have phantom powered mic inputs as well.
You know, I did give REW a shot last night, but I couldn't figure out how to create correction filters with it, though that was before I switched from using Airplay to my Benchmark DAC, so perhaps that was the problem. (latency was causing all sorts of problems in REW)
I've only ever used REW for general measurements and manually doing some EQ or adjusting the gain/crossover on a sub before.
REW is sensitive to latency in my experience, so that may be the issue.
As for getting the correction filters out of REW, try file-->export--> export filters as WAV. That should produce something that can be loaded into the convolution DSP module. However, REW doesn't (to my knowledge) do FIR filtering, so you can also literally just manually dial in the various specified filters into JRiver's PEQ module one for one. That has the advantage of skipping all the latency involved in the convolution process, and will work on all sample rates (separate convolution files need to be created for each sample rate).
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I think the Scarlett would work fine for these purposes (I think Glynor has one of it's bigger cousins, and Matt just picked up a Scarlett 18i20), you could also use something like the Steinberg UR22. Before you get too far down that road, though, you had mentioned in another thread that you were thinking about bi-amping, but with a sub. If that's still on the table you should try and find an interface that has enough outputs for that project too. The focusrite 2i4 or 6i6 both have four analog outputs, the 18i20 has eight outputs. The Steinberg UR824 I mentioned in the other thread has 8 outputs as well. All of those interfaces have phantom powered mic inputs as well.
Hmm, that's true. I was of course thinking that if I ended up going down that route I would simply have the avr/dac handle that, but you have the same problem there of the input/output going through separate interfaces. Perhaps this is something which is only an issue with Acourate though. It's definitely not a problem for Dirac when they are recommending a USB mic.
As for getting the correction filters out of REW, try file-->export--> export filters as WAV. That should produce something that can be loaded into the convolution DSP module. However, REW doesn't (to my knowledge) do FIR filtering, so you can also literally just manually dial in the various specified filters into JRiver's PEQ module one for one. That has the advantage of skipping all the latency involved in the convolution process, and will work on all sample rates (separate convolution files need to be created for each sample rate).
Ah, I see - so REW does not have a one-button option where you select your target curve and it just automatically generates a file to use with convolution? (which Audiolense does)
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Ah, I see - so REW does not have a one-button option where you select your target curve and it just automatically generates a file to use with convolution? (which Audiolense does)
It does have automatic filter generation and a one button export, but that's not necessarily the best way to do it. If you use the "match response to target" option in filter tasks in EQ it will automatically generate a series of filters for you. If you use File--> export-->export filters as WAV, it will give you a convolution file with all the filters included.
But all convolution files are limited to a single sampling rate, so any convolution-based solution requires you to generate multiple convolution filters for different sampling rates.
The reason I suggested entering them manually is that REW doesn't do any filtering that requires convolution to perform (it can all be done with IIR filters). So while there's a convenience value in just exporting it and using convolution, you then have to put up with the added latency that using convolution necessarily entails (which can make a/v sync more challenging). Moving the filters over manually is less convenient, but sidesteps that latency and saves you the step of making the filters for each sample rate.
Does that make sense?
By the way, on the subject of why multiple measurements are a necessity, have a look at slides 101-121 of this presentation for an excellent illustration of why automated room correction based on a single measurement is a bad idea: http://www.hypex.nl/docs/papers/AES123BP.pdf. The author, Bruno Putzeys, is an amp, DAC, and DSP system designer who has done some pretty groundbreaking work in the field.
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Well in this case, it's only going to be playing 16/44.1 so I don't have to worry about that, but it's something I'll have to keep in mind in the future.
I'm not sure that I will have a chance to do anything about it until next weekend now, but I'll give REW a shot.
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I had the opportunity to try REW again this morning, and either I'm doing something wrong, or it's not doing a very good job. All the bass is sucked out (even when sitting at the mic position) and there's a whole lot of treble emphasis when using REW to generate EQ filters.
When using the demo of Audiolense without any changes in setup (mic position, playback/recording levels etc.) it's a good improvement over how the speakers normally sound in the room, and I haven't tried anything like measuring from multiple positions yet.
It's difficult to properly test with the way I have things set up right now, but it does seem like Audiolense XO is producing a better sound than Audiolense 2.0, even when I'm just trying to correct stereo speakers. (I guess that's the True Time Domain correction at work?)
I've now got things setup so that I can view the measurements in Audiolense, but I'm just trying to adjust via the Parametric EQ in Media Center as you suggested.
I'm making some progress in flattening the response, but honestly, I don't really know what I'm doing.
The biggest problem is that I have no idea how to visualize the Q parameter. As I understand it, it's the "width" of the filter, but if I put a filter at 90kHz with a Q of 0.1, 1.0, or 10, I have no idea what that's going to cover. Ideally it would tell me that a filter with a Q of 1 is going to cover a 20Hz band. (±10 from the filter point)
Here's a quick example of what's happening so far after maybe 5-10 minutes of switching over to using the parametric EQ. (which is obviously not the same as room correction) These measurements are taken from one of the worse locations in the room, but it seems that EQ'ing from there seems to result in good sound from most positions. Obviously there's still a lot of work to be done.
(http://www.abload.de/thumb/beforedlf0p.png) (http://www.abload.de/img/beforedlf0p.png) (http://www.abload.de/thumb/afterimfqc.png) (http://www.abload.de/img/afterimfqc.png)
I feel like by the time I'm finished, I'll have a list of 100 filters though with the way things are going so far. :-\
EDIT: After spending a while with this, I think I'm just going to leave it until I have a better space, better equipment, and a better idea of which package it is that I want to buy.
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I had the opportunity to try REW again this morning, and either I'm doing something wrong, or it's not doing a very good job. All the bass is sucked out (even when sitting at the mic position) and there's a whole lot of treble emphasis when using REW to generate EQ filters.
It sounds like it's doing something goofy; How does the measurement look when you measure it once you've dialed in the REW filters?
When using the demo of Audiolense without any changes in setup (mic position, playback/recording levels etc.) it's a good improvement over how the speakers normally sound in the room, and I haven't tried anything like measuring from multiple positions yet.
It's difficult to properly test with the way I have things set up right now, but it does seem like Audiolense XO is producing a better sound than Audiolense 2.0, even when I'm just trying to correct stereo speakers. (I guess that's the True Time Domain correction at work?)
Probably so, the XO has a lot more functionality than the 2.0 as I understand it.
The biggest problem is that I have no idea how to visualize the Q parameter. As I understand it, it's the "width" of the filter, but if I put a filter at 90kHz with a Q of 0.1, 1.0, or 10, I have no idea what that's going to cover. Ideally it would tell me that a filter with a Q of 1 is going to cover a 20Hz band. (±10 from the filter point)
1.4 Q is one octave wide, .66 Q is two octaves wide. Higher is narrower, lower is wider. A 1 Q filter is about 1.4 octaves. For example, a filter at 90 Hz with a 1 Q will mostly affect frequencies between about 35Hz and 250 Hz, but a filter at 40 Hz with a 1 Q will mostly affect frequencies between about 16 Hz and 115Hz. I'd recommend you download some free EQ software that has a GUI to help you visualize. I use the freeware program RePhase: http://sourceforge.net/projects/rephase/ . It's primary function is to generate custom convolution filters, but it has a nice, clean graphical interface that will let you dial things in and show you what effect your EQ will have on frequency and phase. I use it as much for its GUI as for its convolution filter generating capabilities.
Here's a quick example of what's happening so far after maybe 5-10 minutes of switching over to using the parametric EQ. (which is obviously not the same as room correction) These measurements are taken from one of the worse locations in the room, but it seems that EQ'ing from there seems to result in good sound from most positions. Obviously there's still a lot of work to be done.
(http://www.abload.de/thumb/beforedlf0p.png) (http://www.abload.de/img/beforedlf0p.png) (http://www.abload.de/thumb/afterimfqc.png) (http://www.abload.de/img/afterimfqc.png)
I feel like by the time I'm finished, I'll have a list of 100 filters though with the way things are going so far. :-\
That's a definite improvement, no bones about it. Are the measurements similarly improved in other locations? If so, you've already improved your system performance pretty dramatically IMO.
I know what you mean about the filters; they have a tendency to multiply. But you can't EQ every narrow peak and trough (without compromising performance elsewhere in the room), and you'll learn how to consolidate as you go (often one or two filters can take the place of several once you know what the outcome is "supposed" to be).
BTW, your speakers don't happen to be about 2 feet in front of a wall do they?
EDIT: After spending a while with this, I think I'm just going to leave it until I have a better space, better equipment, and a better idea of which package it is that I want to buy.
Well, you've accomplished quite a bit already, but I understand wanting to wait until things are in a more "final" form. If you do decide to press on, it's the sort of thing that works better a little bit at a time. Measurement and correction is impossibly fiddly, and during a long test session our ears/stamina of auditory concentration start to wear out and (at least, in my case) so does my patience ;D
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It sounds like it's doing something goofy; How does the measurement look when you measure it once you've dialed in the REW filters?
That was before I installed ASIO Bridge (http://vb-audio.pagesperso-orange.fr/Cable/index.htm) to route the sound through Media Center's DSP engine, so I didn't get a chance to measure it. By the time I set all that up, I had given up on REW and uninstalled Java. (I really don't want to leave Java on my system) Maybe I'll give it another look next weekend, but I don't know that I have the willpower to spend another afternoon doing this.
1.4 Q is one octave wide, .66 Q is two octaves wide. Higher is narrower, lower is wider. A 1 Q is about 1.4 octaves. For example, a filter at 90 Hz with a 1 Q will mostly affect frequencies between about 35Hz and 250 Hz, but a filter at 40 Hz with a 1 Q will mostly affect frequencies between about 16 Hz and 115Hz. I'd recommend you download some free EQ software that has a GUI to help you visualize. I use the freeware program RePhase: http://sourceforge.net/projects/rephase/ . It's primary function is to generate custom convolution filters, but it has a nice, clean graphical interface that will let you dial things in and show you what effect your EQ will have on frequency and phase. I use it as much for its GUI as for its convolution filter generating capabilities.
Thanks, I just saw your write-up on the parametric EQ, so I have a better idea of it now. I suppose the thing I don't understand is why "Q" exists at all. I mean, I know that it makes sense from an engineer's point of view, but why not just tell me the range a filter is going to affect? So if I put a filter at 90Hz with a Q of 1, just tell me that it's going to affect 35-250Hz rather than assume I know what Q means.
That's a definite improvement, no bones about it. Are the measurements similarly improved in other locations? If so, you've already improved your system performance pretty dramatically IMO.
Yes, it was a similar improvement in most places. I think the problem is that once you start doing very fine-grained adjustments, it becomes a lot more location specific.
What I suppose I ought to do is measure a lot of the possible listening locations in the room, average them, and find a spot which measures close to the average and work off that.
I spent a bit of time after that getting it looking quite a bit flatter still, but being careful not to smooth out the peaks/troughs too much, so that you ended up with a very uneven response across the room. It's sounding good, but not as good as the results from Audiolense XO. (and presumably what Acourate or Dirac could do as well)
Well, you've accomplished quite a bit already, but I understand wanting to wait until things are in a more "final" form. If you do decide to press on, it's the sort of thing that works better a little bit at a time. Measurement and correction is impossibly fiddly, and during a long test session our ears/stamina of auditory concentration start to wear out and (at least, in my case) so does my patience ;D
I only had so much time to spend with it today, and it's really something where spending a lot of time on it feels like it's a task which could just be completely automated... which I suppose is what the products like Audiolense are doing.
That said, I do know from experience that calibrating a display by hand often results in better performance than completely automated calibration, so I suppose it's the same thing there.
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Thanks, I just saw your write-up on the parametric EQ, so I have a better idea of it now. I suppose the thing I don't understand is why "Q" exists at all. I mean, I know that it makes sense from an engineer's point of view, but why not just tell me the range a filter is going to affect? So if I put a filter at 90Hz with a Q of 1, just tell me that it's going to affect 35-250Hz rather than assume I know what Q means.
I agree that it would be easier to read that way for sure. The filters are emulating analog filters, and Q is one of the mathematical parameters that define the filters in analog. Also (and this part is a headache) the higher the amplitude of the filter the wider the range it will have a meaningful effect on. The majority of the effect will be in the range I described for any amplitude, but, say, a 10 dB boost will have a very "long tail." I've attached two example filters from RePhase. Both are 1Q 90Hz filters, one is 3dB, one is 6dB.
Yes, it was a similar improvement in most places. I think the problem is that once you start doing very fine-grained adjustments, it becomes a lot more location specific.
Yes. Most of those small dips and peaks are the result of room echoes and cancellations, and they'll differ everywhere throughout the room.
What I suppose I ought to do is measure a lot of the possible listening locations in the room, average them, and find a spot which measures close to the average and work off that.
That's a good idea. One other tip: when you're sitting back down with this again, try measuring very close to the speaker for an additional data point. Try at one meter (or slightly closer) with the mic positioned between the two drivers. Then try measuring with the mic directly in front of the mid-bass element, about 2 or 3 inches away. The latter measurement won't give you any useful information about the treble region, but will give you a better idea of what your speakers natural bass response is (at that distance the direct output will tend to dominate the room response). Better would be to conduct those kinds of measurements outside, or, barring that, temporarily move your speakers farther away from the walls of the room they're in for measurement purposes (at least five or six feet if possible). I'd guess your speakers are about two or two and a half feet from the back wall right now based on the measurements?
Generally, you can safely EQ any peaks or dips that result from the speaker's response, it's the room that's harder to fix in EQ.
I spent a bit of time after that getting it looking quite a bit flatter still, but being careful not to smooth out the peaks/troughs too much, so that you ended up with a very uneven response across the room. It's sounding good, but not as good as the results from Audiolense XO. (and presumably what Acourate or Dirac could do as well)
I only had so much time to spend with it today, and it's really something where spending a lot of time on it feels like it's a task which could just be completely automated... which I suppose is what the products like Audiolense are doing.
That said, I do know from experience that calibrating a display by hand often results in better performance than completely automated calibration, so I suppose it's the same thing there.
Some parts can potentially be automated (calculating the filters, applying them, etc.); other parts can't be automated (learning measurement technique, understanding where the automated tools may overreach, etc.), so you're not wasting your time learning about the process, and it sounds like you're getting some improvements in system performance into the bargain.
Even with a theoretically perfect fully automated solution there's also always room for tweaking its target for individual taste, your own ears, etc. For example, many people don't prefer the sound of a flat frequency response, they prefer a gradual roll down across the frequency band from bass to treble (sometimes called a house curve or x-curve). Alternatively, I think Bob Katz opined on the forums that he likes it flat to 1 or 2 KHz and then gradually attenuating to about 10 dB down at 20KHz.
Even if you wind up going with Audiolense, there'll probably still be some room for tweaking.
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I know that I've only mentioned Acourate, Audiolense, and Dirac, but that's because it's all I'm aware of. I've not ruled out anything else if there's other software you would recommend instead.
There is also this open source drc solution aptly named drc.
http://drc-fir.sourceforge.net (http://drc-fir.sourceforge.net)
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I was wondering if anyone here had advice or recommendations about room correction.
I know that I've only mentioned Acourate, Audiolense, and Dirac, but that's because it's all I'm aware of. I've not ruled out anything else if there's other software you would recommend instead.
I have tried all of the software mentioned in this thread. On my system, I have had the best results with Acourate. I wrote a how to article here: http://www.computeraudiophile.com/content/529-acourate-digital-room-and-loudspeaker-correction-software-walkthrough/
There is a specific target response that offers a perceptually flat frequency response at the listening position, that is covered off in the article. There seems to be some consensus on this as various people have tried in several different systems and rooms and consistently get the desired result.
I used to use REW, but the issue is that, for room correction, REW does not use frequency dependent windowing (FDW) and therefore the result does not match what our ears hear. At least that has been my experience.
Another good website is: http://digitalroomcorrection.hk/http___www.digitalroomcorrection.hk_/Welcome.html and this is a nice article: http://www.acourate.com/freedownload/TonyKnightSystemDescription.pdf .
Hope that helps. Cheers, Mitch
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I have tried all of the software mentioned in this thread. On my system, I have had the best results with Acourate. I wrote a how to article here: http://www.computeraudiophile.com/content/529-acourate-digital-room-and-loudspeaker-correction-software-walkthrough/
There is a specific target response that offers a perceptually flat frequency response at the listening position, that is covered off in the article. There seems to be some consensus on this as various people have tried in several different systems and rooms and consistently get the desired result.
I used to use REW, but the issue is that, for room correction, REW does not use frequency dependent windowing (FDW) and therefore the result does not match what our ears hear. At least that has been my experience.
Another good website is: http://digitalroomcorrection.hk/http___www.digitalroomcorrection.hk_/Welcome.html and this is a nice article: http://www.acourate.com/freedownload/TonyKnightSystemDescription.pdf .
Hope that helps. Cheers, Mitch
Thank you for the links and recommendation - I read your article yesterday and found it very informative. I also tried using that curve in the Audiolense trial.
It's a shame that Acourate does not have a fully functional demo. You can send in your measurements and a couple of tracks to receive processed tracks, but that's not nearly as convenient as being able to generate them in the demo when there are so many options to explore.
As you are a user of Acourate, can you tell me if it lets you take measurements from multiple listening positions when building a correction?
Ideally you would treat the room to even out the response rather than use digital correction, but that's not really an option here.
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Thank you for the links and recommendation - I read your article yesterday and found it very informative. I also tried using that curve in the Audiolense trial.
It's a shame that Acourate does not have a fully functional demo. You can send in your measurements and a couple of tracks to receive processed tracks, but that's not nearly as convenient as being able to generate them in the demo when there are so many options to explore.
As you are a user of Acourate, can you tell me if it lets you take measurements from multiple listening positions when building a correction?
Ideally you would treat the room to even out the response rather than use digital correction, but that's not really an option here.
I mention the target curve as other DRC software I have tried seemed to have different ways of interpreting the curve. In one, the correction seemed to either overshoot or undershoot the target, no matter what I did. In Acourate, the first time was the last time.
Wrt to measuring multiple listening positions, I did try that in other DRC software and found, at least in my setup, that taking measurements in multiple listening positions did not produce a better corrected response. The best results I got was to triangulate on the listening position/speakers and take one measurement and let the speakers natural polar response deal with various other listening positions. With frequency domain windowing, the window is open in the 100's of milliseconds in the low end and less than a millisecond at the very top. With a FIR filter length of 65,536 taps and 100's of milliseconds of correction in the low end, the bottom end sounds good in my listening room no matter where I sit.
To answer the question, there is not a ready to go function like that in Acourate as far as I know, but I did not look for it as I don't require it. But I would not be surprised if it could be done as Acourate is more of a digital audio toolbox with many possibilities. If really interested, you may want to ask in the Acourate forum: http://groups.yahoo.com/neo/groups/acourate/info
Yes, I still use Acourate and wrote this advanced article. Hopefully I am not spamming and folks find it useful:
http://www.computeraudiophile.com/content/556-advanced-acourate-digital-xo-time-alignment-driver-linearization-walkthrough/
Cheers, Mitch
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I mention the target curve as other DRC software I have tried seemed to have different ways of interpreting the curve. In one, the correction seemed to either overshoot or undershoot the target, no matter what I did. In Acourate, the first time was the last time.
Wrt to measuring multiple listening positions, I did try that in other DRC software and found, at least in my setup, that taking measurements in multiple listening positions did not produce a better corrected response. The best results I got was to triangulate on the listening position/speakers and take one measurement and let the speakers natural polar response deal with various other listening positions. With frequency domain windowing, the window is open in the 100's of milliseconds in the low end and less than a millisecond at the very top. With a FIR filter length of 65,536 taps and 100's of milliseconds of correction in the low end, the bottom end sounds good in my listening room no matter where I sit.
To answer the question, there is not a ready to go function like that in Acourate as far as I know, but I did not look for it as I don't require it. But I would not be surprised if it could be done as Acourate is more of a digital audio toolbox with many possibilities. If really interested, you may want to ask in the Acourate forum: http://groups.yahoo.com/neo/groups/acourate/info
Yes, I still use Acourate and wrote this advanced article. Hopefully I am not spamming and folks find it useful:
http://www.computeraudiophile.com/content/556-advanced-acourate-digital-xo-time-alignment-driver-linearization-walkthrough/
Cheers, Mitch
That really is an impressive article Mitchco. I read it through a few times last night, and even though I don't use Acourate, I definitely learned a thing or two :) As I was reading it, it occurred to me that, while Acourate has some pretty unique functionality, some of the things you're doing for your system with Acourate could potentially be done manually with JRiver and free measurement software (like Holm Impulse, which also offers frequency dependent windowing). Obviously Acourate can do things that would be much harder (or impossible) to do without proprietary software, but even folks who don't use Acourate could definitely benefit from reading your article and digesting the underlying principles. There's a lot of good acoustic theory in there.
Which all got me thinking: would there be any interest around here in something similar to Mitchco's article, except focused on crossover and speaker optimization that can be accomplished using only JRiver and freeware? Obviously it won't get you quite as far, but can get you a good distance. I say that because I'm getting ready to start work on a pair of bi-amped bookshelf speakers that I plan to finish putting together and tuning over the next month or two. Because I'll be going through the steps anyway, I could document the measurements/process as I go.
And I hope I'm not stepping on your toes Mitchco, I just found your article very inspiring ;D
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That really is an impressive article Mitchco. I read it through a few times last night, and even though I don't use Acourate, I definitely learned a thing or two :) As I was reading it, it occurred to me that, while Acourate has some pretty unique functionality, some of the things you're doing for your system with Acourate could potentially be done manually with JRiver and free measurement software (like Holm Impulse, which also offers frequency dependent windowing). Obviously Acourate can do things that would be much harder (or impossible) to do without proprietary software, but even folks who don't use Acourate could definitely benefit from reading your article and digesting the underlying principles. There's a lot of good acoustic theory in there.
Which all got me thinking: would there be any interest around here in something similar to Mitchco's article, except focused on crossover and speaker optimization that can be accomplished using only JRiver and freeware? Obviously it won't get you quite as far, but can get you a good distance. I say that because I'm getting ready to start work on a pair of bi-amped bookshelf speakers that I plan to finish putting together and tuning over the next month or two. Because I'll be going through the steps anyway, I could document the measurements/process as I go.
And I hope I'm not stepping on your toes Mitchco, I just found your article very inspiring ;D
Thanks and go for it! I for one would find it interesting and useful as I was going to try Holm Impulse and RePhase, but could not find any docs on how to actually use it. Many would be grateful for your efforts would be my take.
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Thanks and go for it! I for one would find it interesting and useful as I was going to try Holm Impulse and RePhase, but could not find any docs on how to actually use it. Many would be grateful for your efforts would be my take.
I know what you mean; the Holm documentation is pretty lacking, and the discussions over at the "official" threads often shed a little more heat than light on the issues. There are a few RePhase tutorials out there, but the good ones just happen to be in French :-\
I'll see if I can't get my ducks in a row over the holidays.
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My problem is I can't really can't do any "physical" room correction .... it is the worst room imaginable for audio - one wall is glass, different ceiling height from 2.5 meters to almost 8 meters. Floor in stone tiles - no rugs possible. Speaker placement atrocious. (no choice without jackhammering my fireplace to pieces) So all this is very interesting. Thanks Mitchco
would there be any interest around here in something similar to Mitchco's article, except focused on crossover and speaker optimization that can be accomplished using only JRiver and freeware?
Yes very much so! Can't swing buying acourate, and even with a good guide, the learning curve looks steep. I can't adjust my active analog Xos really
Really thinking of doing digital active crossovers. And I like "free"! (Would jriver be providing microphones like this (http://www.earthworksaudio.com/microphones/qtc-series-2/qtc40/) at the same price as a their remote ?) ;D
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I've spent some time looking over hardware again today, and I'm still not entirely clear on what to get. As far as software is concerned, I think I'm convinced that Acourate is what I want to get now.
I don't think there is anywhere here selling calibrated USB mics though - my only option would be to have that Dayton OmniMic shipped over from the US, and I'm not clear on what the shipping charges would be. (I hate that websites require you to register first)
I'm somewhat hesitant to go with a USB mic now though, as I've seen a lot of reports about high noise floors with them (though nothing about the OmniMic) and that they operate at a fixed gain so that measurements have to be very loud to get a good signal level.
Alternatively, there are a few calibrated phantom power mics available here, and buying one of them and a Scarlett 2i2 or Steinberg UR22 is likely going to come to around the same price as the OmniMic plus shipping and import fees. I don't know that either the 2i2 or UR22 have ASIO drivers though, so there's a chance I would still have to go through ASIO4All anyway. (Acourate only works with ASIO devices) This would also mean that I'm probably spending as much on the interface as the mic, rather than it all going to the mic.
While I am still hesitant, it does seem that using ASIO4All should not be a problem with Acourate though:
Indeed it is the only chance to use two different devices for playback and recording by combining them with Asio4All.
This will work but some attention is required regarding clocks. If two clocks are in game for playback and recording (e.g. USB mic) then they will not fit together by high chance. But it is possible to compensate for this. The Acourate logsweep recorder allows to add Dirac pulses to a sweep. The distance is known. So if the recording shows another distance it is possible to create an inverse logsweep (required for pulse response calculation) with the correct length. Thus the pulses will be correct
But I'm unsure about whether this is something that will have to be done for each measurement, which would be a major nuisance (latency in ASIO4All seems variable) or if it's just something that has to be done once.
With display calibration, some of my hardware requires a dark calibration every 10-20 minutes, and that sort of thing quickly becomes very frustrating.
Should I maybe look into battery-powered Mics, or using external amplifiers, and simply buying a PCIe sound card that has an ASIO driver?
I'd really prefer to be using a USB device rather than PCIe though - that way I'm future-proofed if I ever wanted to use a laptop for measurements.
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I've spent some time looking over hardware again today, and I'm still not entirely clear on what to get. As far as software is concerned, I think I'm convinced that Acourate is what I want to get now.
I don't think there is anywhere here selling calibrated USB mics though - my only option would be to have that Dayton OmniMic shipped over from the US, and I'm not clear on what the shipping charges would be. (I hate that websites require you to register first)
I'm somewhat hesitant to go with a USB mic now though, as I've seen a lot of reports about high noise floors with them (though nothing about the OmniMic) and that they operate at a fixed gain so that measurements have to be very loud to get a good signal level.
The noisefloor on mine is fairly low. I haven't measured it successfully, but I haven't run up against it in my measurements (the noise floor in my room is higher than the noise floor on the mic). It is fixed gain, but I can tell you that I have to turn the windows audio input levels way down to avoid overloading it. Low sensitivity is not the issue with the Omnimic. But it isn't the cheapest solution, and if you're definitely planning to do Acourate, you might want to go phantom powered with an external interface.
Alternatively, there are a few calibrated phantom power mics available here, and buying one of them and a Scarlett 2i2 or Steinberg UR22 is likely going to come to around the same price as the OmniMic plus shipping and import fees. I don't know that either the 2i2 or UR22 have ASIO drivers though, so there's a chance I would still have to go through ASIO4All anyway. (Acourate only works with ASIO devices) This would also mean that I'm probably spending as much on the interface as the mic, rather than it all going to the mic.
The UR22 definitely has an ASIO driver (it's discussed in the manual http://download.steinberg.net/downloads_hardware/UR22/UR22_documentation/UR22_OperationManual_en.pdf). Steinberg actually maintains the ASIO specification, so I'd be shocked if it didn't. I haven't used the UR22, but the UR824 ASIO driver has been completely frictionless for me. Maybe somebody with a Scarlett can comment as well?
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Any pro audio device will have an ASIO driver. I own the Steinberg UR824, but I also have owned several Tascam products and currently have a Tascam US-366. It is 4 channels so can be used as a preamp and for a 2 channel system with 1 or 2 subwoofers. If you only need a preamp or 2 channels, then I would recommend the Tascam US-122MKII. I've used one quite a bit with REW.
I own a Behringer ECM8000 microphone (same as sold by Acourate) but my current measurement microphone is an iSEMcon EMM-7101-CHTB. It is much more accurate, lower noise floor, and flat from 5 Hz to 20 kHz. iSEMcon is in Germany and Audiolense sells their MP-1r-KIT measurement kit which includes a preamp. Unless you need to measure below 20 Hz or at high SPL levels, then this would work great. You can buy direct from iSEMcon, too.
If you do get something like the Behringer ECM8000, then it needs to be professionally calibrated. You can load the calibration file in Audiolense or REW. I'm not sure about Acourate. iSEMcon individually calibrates all their mics and includes a calibration file.
I've also used an Omnimic. It is an excellent solution and extremely easy to use. It is very easy to take to someone else's house and you can take a measurement in just a few seconds since it uses playback from a CD. The other programs generate their own signal which makes it a lot more difficult to measure on another system than your own.
I use Audiolense XO and really like it. The system setup, measurement workflow, and filter creation is very nicely laid out in my opinion and makes measuring a 7.1 system with multiple subwoofers extremely easy. You can use Audiolense for free to take measurements, export them, and then import them into REW for filter generation. You get a better measurement than REW for the reason's Mitcho provided, but can still use REW for analysis and filter generation. This is a great free solution, IMO.
I take my measurements at 48 kHz and then generate 48 kHz filters with Audiolense. JRiver will automatically resample the filter to match the current output. I have found this simpler than creating multiple filters for each sample rate. Also, my 48 kHz measurement provides the best convolution results with the lowest noise floor.
Here is how Dirac Live handles different input/output devices:
... as far as I know other DRCs require to connect the mic to the same usb port as the output device to make sure they share the same clock source (so a USB mic on one port and a DAC on another USB port is a problem). But you may have noticed that when you do each stereo measurement with Dirac Live three sweeps are played instead of two as expected...this way you can have completely different clock sources on the playback and recording device because we compensate for the timing problems. This is why we play three sweeps when we do a stereo measurement, first left, then right, then left again. When we do like this we can calculate the clock drift between the playback and recording device and compensate for it.
I started a thread on Audiolense's forum 11/20 asking for similar support. However, if their is variable clock drift then this method still doesn't work.
Mitcho - thanks for your article. I have everything I need to go fully active with my main speakers, but still haven't taken that step.
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You can use Audiolense for free to take measurements, export them, and then import them into REW for filter generation. You get a better measurement than REW for the reason's Mitcho provided, but can still use REW for analysis and filter generation. This is a great free solution, IMO.
That's a great point, I've been doing the same thing with Holm for a while (using it as a measurng tool and then exporting to REW) because Holm allows for frequency dependent windowing and/or gating. It hadn't occurred to me to use Audiolense's measurement suite and do the same thing. Thanks for the tip ;D
I take my measurements at 48 kHz and then generate 48 kHz filters with Audiolense. JRiver will automatically resample the filter to match the current output. I have found this simpler than creating multiple filters for each sample rate.
I'll confess that I never got that feature working right, personally. I know it works for folks, but I never got multiple sample rates working correctly with the same filter (they always sounded mighty odd). Maybe I'll take another run at it and see if I have better luck now.
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Thanks again everyone. At some point, I had been thinking about purchasing an ADC (probably a Benchmark ADC to go along with my DAC) but my requirements for recording are really not going to be that high end.
It looks like one of those Steinberg interfaces would be perfect though, as it will serve as an input/output device for Acourate without any latency problems, and be useful for recording purposes if I buy another mic. (rather than a measurement mic)
The UR824 looks like a great piece of kit, but it's a lot more than I'm wanting to spend right now.
I could stretch to the UR44 which I'd prefer to have due to its wider selection of inputs/outputs and the half-rack size (same as Benchmark) but it requires external power and won't be available until Q1 2014.
The UR22 seems like it would keep things as simple as possible, being completely bus powered. Just a shame it's a 1/3 rack width device.
Of course at some point down the line, I may want to move to a multichannel setup to use a digital crossover, but realistically that's not going to happen any time soon. Perhaps then I will be looking into the UR824 or similar units.
Now I just need to see what mics are available to buy here with individual calibrations. The Behringer ECM8000 is easy enough to get, but there might be some better options that don't cost a lot more.
I'll confess that I never got that feature working right, personally. I know it works for folks, but I never got multiple sample rates working correctly with the same filter (they always sounded mighty odd). Maybe I'll take another run at it and see if I have better luck now.
Is it really that difficult to set up multiple sample rates anyway? It seemed like it would be easy enough to leave everything as it is, switch the sample rate and take another set of measurements.
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Now I just need to see what mics are available to buy here with individual calibrations. The Behringer ECM8000 is easy enough to get, but there might be some better options that don't cost a lot more.
I had a calibrated ECM-8000 before I had an Omnimic. It will work, but it does have a few quirks to get used to (positioning is more important than it should be with a condenser mic, very low frequency sensitivity isn't the best), but it was perfectly satisfactory for my measuring purposes. I only switched to get a slightly less "positional" mic and for the convenience of USB.
Is it really that difficult to set up multiple sample rates anyway? It seemed like it would be easy enough to leave everything as it is, switch the sample rate and take another set of measurements.
No, once you figure out the naming convention/config file writing it's relatively easy. Just repetitive. Although some tools (like REW) don't support all sample rates as outputs, so there's that. And it may not even turn out to be an issue for you, anyway.
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I had a calibrated ECM-8000 before I had an Omnimic. It will work, but it does have a few quirks to get used to (positioning is more important than it should be with a condenser mic, very low frequency sensitivity isn't the best), but it was perfectly satisfactory for my measuring purposes. I only switched to get a slightly less "positional" mic and for the convenience of USB.
It looks like I will probably buy a mic from iSEMcon, as they sell calibrated Behringers and their own mics: http://www.acousticsshop.isemcon.com/index.php?cPath=21_23 (http://www.acousticsshop.isemcon.com/index.php?cPath=21_23)
One of theirs with a UR22 looks like it will cost me about the same as getting an OmniMic shipped over.
While there'll be a few more pieces involved, with the UR22 being bus-powered it should be as easy to use as a dedicated USB mic, and a bit more flexible as I can repurpose the UR22 later.
No, once you figure out the naming convention/config file writing it's relatively easy. Just repetitive. Although some tools (like REW) don't support all sample rates as outputs, so there's that. And it may not even turn out to be an issue for you, anyway.
Well in this case, the speakers will be hooked up to an AirPort Express which only outputs 44.1kHz, so it won't be an issue, but in the future multiple sample rates will matter once I get speakers set up elsewhere.
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Is it really that difficult to set up multiple sample rates anyway? It seemed like it would be easy enough to leave everything as it is, switch the sample rate and take another set of measurements.
You have the following options:
1. Use 1 measurement and have the measurement software create filters for each sample rate. JRiver will automatically switch filters if named correctly.
2. Use 1 measurement, one filter, and have JRiver resample the filter.
3. Use a measurement for each sample rate and a filter for each sample rate.
I choose #2 because I think it sounds better than #1 when I've done comparisons, it is easier to manage all my filters, the hardware and microphone noise floor is lower at 48 kHz than higher sample rates, and I don't have to go through a long manual process of finding the best filter for the measurement when doing multiple sample rates. So far, each measurement is slightly different and seems to require tweaking for the best sound. When doing 6 measurements and 6 filters for a 7.2 surround system it can take a while. Also, I do different filters for different media types so that would be 24-30 filters I would need to create. Now I only need 4-5.
#3 is also dependent on your hardware. You may have 192 kHz output, but the preamp can only record at 96 kHz, for example.
I forgot to mention earlier than I currently just do my filters with a single measurement. I plan to do some multi-seat measurements sometime, but haven't gotten around to it.
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I also use the TASCAM unit per Mojave's recommendation. It's great.
I have used all three software and prefer DIRAC LIVE. I can't say I am expert level like Mojave's is to audiolense or Mitch is to both audiolense and acourate. I can say that right "out of the gate" I get much better sound with DIRAC. I really think Bruno is right that multi measurements need to be done (at least in my room). JJ Johnston also did an AES paper on the same topic and he concluded that a tetrahedron shape works well for DSP measuring.
Michael,
I know DIRAC claims to able to account for clock drift, but not with my crazy hardware. :-) I still use a synchronous measurement. Now I am using my RME card, with a grace m101 and earthworks mic.
Any pro audio device will have an ASIO driver. I own the Steinberg UR824, but I also have owned several Tascam products and currently have a Tascam US-366. It is 4 channels so can be used as a preamp and for a 2 channel system with 1 or 2 subwoofers. If you only need a preamp or 2 channels, then I would recommend the Tascam . . .
I started a thread on Audiolense's forum 11/20 asking for similar support.
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I'm not sure if I've read into this correctly (lots to take in) but I'm concerned about the combination of hardware required to make this stuff work properly.
I've bought this measurement kit: MP-1r-KIT Acoustical measurement kit
My worry is all the mention of latency and using the same input/output device.
My pc has on board sound, but I have an outboard dac (same as most here, I'd imagine) which I feed a signal via asynchronous usb.
Will this be a problem with Dirac/Acourate?
*I know that I plug the mic into the audio input on my pc's motherboard,
*get the software to make it's noises and take the measurements
*create the corrections with whichever program I choose
*and apply the convolution to JRiver
Isthat right?
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I am told it won't matter with DIRAC. But I use a very unusual DAC so it matters with my DAC. I would say it probably doesn't matter with your DAC. With DIRAC you can use a different input and output device. That's a good measurement kit. If I were you, I wouldn't use the mobo mic. I have no experience there but I doubt you will be happy with it. I would get a single device to go out and in like the TASCAM models previously mentioned. That's a much cleaner solution. All of the software mentioned are pretty straightforward to use. However , the filter generated can only be as good as the measurement. GIGO.
I'm not sure if I've read into this correctly (lots to take in) but I'm concerned about the combination of hardware required to make this stuff work properly.
I've bought this measurement kit: MP-1r-KIT Acoustical measurement kit
My worry is all the mention of latency and using the same input/output device.
My pc has on board sound, but I have an outboard dac (same as most here, I'd imagine) which I feed a signal via asynchronous usb.
Will this be a problem with Dirac/Acourate?
*I know that I plug the mic into the audio input on my pc's motherboard,
*get the software to make it's noises and take the measurements
*create the corrections with whichever program I choose
*and apply the convolution to JRiver
Isthat right?
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Now I just need to see what mics are available to buy here with individual calibrations..
I'd suggest getting a mic from Herb at Cross Spectrum Labs. He calibrates Dayton mics to his professional reference. You get multiple calibration files (for using the mic at different angles) on a thumb drive and printouts of the mic's response and calibration. Good prices and I think he offers the usb mic as well.
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I'm not sure if I've read into this correctly (lots to take in) but I'm concerned about the combination of hardware required to make this stuff work properly.
I've bought this measurement kit: MP-1r-KIT Acoustical measurement kit
My worry is all the mention of latency and using the same input/output device.
My pc has on board sound, but I have an outboard dac (same as most here, I'd imagine) which I feed a signal via asynchronous usb.
Will this be a problem with Dirac/Acourate?
*I know that I plug the mic into the audio input on my pc's motherboard,
*get the software to make it's noises and take the measurements
*create the corrections with whichever program I choose
*and apply the convolution to JRiver
Isthat right?
Dirac has a method of matching up the timing from an input that is distinct from an output (although it sounds like it doesn't work correctly with all hardware, based on dallasjustice's experience). According to info upthread, it sounds like even Acourate (which only supports ASIO) would work with two different devices with ASIO4All. So it sounds like most of this software can be made to work with two different devices, it's just easier to get a predictable result if the same device is used for in and out.
The underlying technical issue is that these programs attempt to make complex and minute corrections in the time domain, by altering the timing of various frequencies (i.e. "phase"). In order to make those kinds of delicate timing corrections successfully, software needs to know exactly (to within microseconds) when the test signal "starts" in the recording. The problem is that different audio devices introduce different amounts of delay as part of their operation, and the amount of delay introduced may not be constant even for a single device over time. At minimum, to use two different devices, software needs a way to determine what the true "start time" of the recorded signal is and potentially a way to deal with "clock drift" during the recording itself.
The first issue is easier to fix than the second one. Any software that records an impulse response *should* be able to identify when the "start" of an impulse is (The start of an impulse is obvious enough in most cases that it could be identified manually with pretty good precision, and free measurement suites like Holm do it automatically). If the software you're planning to use doesn't explicitly support using two devices, check and see if it a) takes impulse measurements and b) offers a way to adjust or alter the "time zero." If both of those options are present it should be possible to work around this issue.
The second problem (clock drift during measurement) is harder to get a handle on, but given the length of most log sweeps (ten to twenty seconds), it may or may not be a huge issue (especially if you don't try to do phase correction at higher frequencies, i.e. late in the logsweep when the clock may have drifted by a significant amount). But I've seen some pretty impressive clock drifts between devices too, so it's hard to speak in generalities. This hasn't been a problem for me in my own measurements using two different devices, but it might be a problem for some hardware. The only way to know about the second issue, for sure, with a given set of hardware is to measure and see if there are phase anomalies in your measurement. I would expect to see a uniform, gradual rise or fall in phase over the course of the measurement which is not correlated to changes in frequency response. I'd advise taking some measurements with your setup and see if the phase looks how you'd expect.
So while there are ways to work around it, the safest course (unless you're sure the software you plan to use supports using two different devices) is to get one of the audio interfaces mentioned upthread that can handle both input and output (unless you enjoy tinkering with measurements). It will probably save you some heartburn down the road.
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I'd suggest getting a mic from Herb at Cross Spectrum Labs. He calibrates Dayton mics to his professional reference. You get multiple calibration files (for using the mic at different angles) on a thumb drive and printouts of the mic's response and calibration. Good prices and I think he offers the usb mic as well.
Thanks for the recommendation, it does look like getting a mic shipped over from the US may be my best option.
I've spent a lot of time reading about mics now, and I'm thinking I may stretch to a SF101a microphone (http://www.testmic.com/index.html#mics).
It's a lot more than I was planning on spending, but it seems like it would be worthwhile for the titanium diaphragm rather than plastic.
I have a lot of display calibration gear, and anything which used plastic filters has lost its accuracy over time (useful once profiled though) but the meters which use glass filters are still accurate and reliable.
Considering that I only plan on making this purchase once, and intend on keeping it for a long time, it seems like it would be worth paying more initially to avoid having to buy another Mic some time in the future.
This was probably not the best time of year to start looking into buying all this equipment, so I'm probably going to hold off on making a decision until early next year now.
If I decide not to buy the SF101a, the Audix TM1 Plus seems like a good choice - though the price for one of them here is much closer to the SF101a than in the US where it's less than half the cost - here it's more like 2/3. Either that or an Earthworks M23, which is similar in price to the TM1+.
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Thanks for the info guys.
Two questions:
1.So a decent mic into my PCs mobo would yield sub optimal results compared to one of these tascam devices? If thats the case, thanks for the info but sheesh, this starts to get expensive quick!
2.Reading the responses to my post might indicate a possible lack of specificity/understanding on my part; when talking about input/output devices.
Input= mic in (analogue) there to facilitate the room measurement work, regardless of software/solution.
Output= USB out (digital) the digitised version of whatever the mic picked up from the analogue input.
All of the above equates to work done on my PCs internal sound, or, one device.
When it hits my (seperate) dac via the USB, that count (to me) as another device.
Is this where the difficulties might occur, as the measurements and convolution/correction are done on the pc and sent to the dac?
I'm not sure if I'm over complicating things; in hindsight possibly yes as all the 'digital' stuff is done on this pc and then packaged and sent to the dac. I also imagine that most people using this stuff have decent, separate dacs to their PCs?
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Thanks for the info guys.
Two questions:
1.So a decent mic into my PCs mobo would yield sub optimal results compared to one of these tascam devices? If thats the case, thanks for the info but sheesh, this starts to get expensive quick!
2.Reading the responses to my post might indicate a possible lack of specificity/understanding on my part; when talking about input/output devices.
Input= mic in (analogue) there to facilitate the room measurement work, regardless of software/solution.
Output= USB out (digital) the digitised version of whatever the mic picked up from the analogue input.
All of the above equates to work done on my PCs internal sound, or, one device.
When it hits my (seperate) dac via the USB, that count (to me) as another device.
Is this where the difficulties might occur, as the measurements and convolution/correction are done on the pc and sent to the dac?
I'm not sure if I'm over complicating things; in hindsight possibly yes as all the 'digital' stuff is done on this pc and then packaged and sent to the dac. I also imagine that most people using this stuff have decent, separate dacs to their PCs?
The issue with two devices isn't necessarily related to outputting sound after you measure; rather it's about outputting sound to create the measurement to begin with. When you measure using your computer, the computer has to generate and play a test signal through your speakers; you then need to record the sound of the test signal playing on the mic.
Step 1: Software generates and begins to "play" the test signal. In your setup, your computer sends a digital signal via USB to your DAC (bypassing your internal soundcard), and the DAC (after some amount of delay) converts the digital audio stream to analog, and sends the now analog audio signal to your amp, and your speakers then play the test signal. There is some amount of time between when you press the "measure" button in whatever software you are using and when the test signal is actually played by your speaker (latency)
Step 2: The sound must travel from your speaker to the microphone. This is not latency, but is actually potentially part of what you're trying to measure.
Step 3: Your microphone picks up the sound of your speakers playing the test signal and then either a) converts the analog signal to digital itself (in the case of a USB mic) or relays the analog measurement back to an analog to digital converter (ADC) in a sound card (in your case the onboard sound card's mic input). In either case the A to D conversion involves some amount of latency, and once the analog signal is digitized it must be relayed from the audio device/driver to the software suite conducting the measurement, which also involves some amount of latency.
If you use only one audio device for both output and input, both the DAC and ADC stage are running off of the same device clock, and are using the same audio driver to communicate with the program. This means that the latency of both step 1 and step 3 are more likely to be similar or the same, and clock drift should not be an issue. If you use a different device for the output of the test signal than for the recording (which it sounds like you are), the latency introduced in Step 1 and Step 3 will be different (different device, different driver), and the ADC may convert the audio at a very slightly different rate than the DAC converted the digital to analog on the front end (different clock).
If software supports using two different devices, it has some mechanism for addressing these issues. If the software doesn't support using two different devices, correcting the different latencies is potentially possible, correcting different device clocks may be much harder (as described above), but the best way to find out if these will be a problem in your software/set up is to try and see.
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The TASCAM us-322 can been purchased new on Amazon for $135. I bet you paid more than that for the mic and pre. You can really only try DIRAC with separate in and out devices. If I were you, I would try all 3 of them. I prefer DIRAC but you may like one of the others better. IME, the benefit is tremendous, if done right.
Thanks for the info guys.
Two questions:
1.So a decent mic into my PCs mobo would yield sub optimal results compared to one of these tascam devices? If thats the case, thanks for the info but sheesh, this starts to get expensive quick!
2.Reading the responses to my post might indicate a possible lack of specificity/understanding on my part; when talking about input/output devices.
Input= mic in (analogue) there to facilitate the room measurement work, regardless of software/solution.
Output= USB out (digital) the digitised version of whatever the mic picked up from the analogue input.
All of the above equates to work done on my PCs internal sound, or, one device.
When it hits my (seperate) dac via the USB, that count (to me) as another device.
Is this where the difficulties might occur, as the measurements and convolution/correction are done on the pc and sent to the dac?
I'm not sure if I'm over complicating things; in hindsight possibly yes as all the 'digital' stuff is done on this pc and then packaged and sent to the dac. I also imagine that most people using this stuff have decent, separate dacs to their PCs?
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Thanks for the clarification Mwillems; youre right, I'll have to have a play and see what happens.
Cheers Dallas, yes I paid a fair amount more than what you've linked for me; I'll have a go with what I've got for and see how it works for me. And I'm pretty excited to see what it's going to do for me! I can only imagine the improvement it will yield (my rooms less than optimal for audio).
I'll report back once I've had a shot with it all.
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The TASCAM us-322 can been purchased new on Amazon for $135. I bet you paid more than that for the mic and pre. You can really only try DIRAC with separate in and out devices. If I were you, I would try all 3 of them. I prefer DIRAC but you may like one of the others better. IME, the benefit is tremendous, if done right.
I was going to say something similar; by the time you're spending several hundreds of euros for one of these software packages, another $100 or so for an external interface doesn't seem like a huge investment.
Also, he's not necessarily limited to DIRAC. It took a little fiddling, but I used two different devices and managed to get good measurements using the Audiolense trial. I made two sets of measurements: one with a microphone on the same device I was using for output (as a control), and then made one with a USB mic instead, and managed to get virtually identical measurements after some paddling around/reconfiguration.
It's obviously not recommended (the software gave me dire warnings when I had to turn off ASIO), and I wouldn't advise someone starting from scratch to try and use two different devices for software that doesn't officially support it. But different hardware will produce different results (which is the whole problem), and if you've already got the hardware, it's free to try the measurement part of Audiolense and see if you get flaky results with two devices.
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@Mwillems, I'm sorry I'm really new to this side of audio.
You're agreeing with Dallas by saying the tascam is superior to the mic input on a mobo for taking measurements?
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@Mwillems, I'm sorry I'm really new to this side of audio.
You're agreeing with Dallas by saying the tascam is superior to the mic input on a mobo for taking measurements?
Yes. Almost any sound device will be superior to Mobo sound, and having one device for both input and output during measurement will make your life easier.
The extra device will cost you something, but most of the software packages being discussed here are themselves pretty expensive (between $250 and $650).
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I understand with the kit I have it works like this:
Mic > mic pre > mobo (via 3.5mm jack) = recording
I'm not sure I understand how the tascam would fit into the equation, I can only guess it would look like this:
Mic > tascam pre > mobo (via USB) = recording
And the benefit of the tascam is not using the headphone input?
Thanks for your explanation!
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I understand with the kit I have it works like this:
Mic > mic pre > mobo (via 3.5mm jack) = recording
I'm not sure I understand how the tascam would fit into the equation, I can only guess it would look like this:
Mic > tascam pre > mobo (via USB) = recording
And the benefit of the tascam is not using the headphone input?
Thanks for your explanation!
That's about right, but I wanted to clarify one thing. The key difference between those two signal chains is that the TASCAM would be handling the Analog to Digital conversion (ADC) and sending a digital signal to the computer via USB, instead of your current setup which sends an analog signal to the analog to digital converter built into the motherboard (the MoBo jack). Think about the quality of the DAC on the MoBo soundcard as compared to the outboard DAC you have. The ADC in the MoBo soundcard is the same way, an outboard ADC is likely to be much better.
The other advantage is that, if you pick an external device with both a DAC and an ADC in it (like the TASCAM), you can use it to handle both sides of the measurement process (and switch back to your existing DAC when you're done measuring).
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The tascam also can go out as well. So, you plug the tascam into the usb. The tascam has a mixer which also has a control panel that allows you to configure it to send 2 channels out the back of the tascam into your amp while also using the mic input to go into the tascam. This way the output and the input are controlled by the clock inside the tascam unit. Thus, the measurement will be synchronous.
One word of caution: I have done this configuration. If you go directly to your amp out of the tascam, make sure you know how the volume control works in the tascam mixer. I like to use a couple of 12db resistors between the tascam and my amps. That way, I can confirm the volume is working correctly before I send a test signal through. I have blown a tweeter in alll of this. :-) I had to send my speaker back to the manufacturer to have the tweeter replaced and remeasured. Kinda takes the fun out of it for a short time. ;D
I understand with the kit I have it works like this:
Mic > mic pre > mobo (via 3.5mm jack) = recording
I'm not sure I understand how the tascam would fit into the equation, I can only guess it would look like this:
Mic > tascam pre > mobo (via USB) = recording
And the benefit of the tascam is not using the headphone input?
Thanks for your explanation!
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One word of caution: I have done this configuration. If you go directly to your amp out of the tascam, make sure you know how the volume control works in the tascam mixer. I like to use a couple of 12db resistors between the tascam and my amps. That way, I can confirm the volume is working correctly before I send a test signal through. I have blown a tweeter in alll of this. :-) I had to send my speaker back to the manufacturer to have the tweeter replaced and remeasured. Kinda takes the fun out of it for a short time. ;D
Yikes, that's no fun. Tweeters are delicate creatures. I've been lucky so far, but there were one or two episodes where I was sure I'd fried them (but thankfully hadn't).
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Thank you both for your input and information, gentlemen.
When you explain about the tascam as a superior ADC, it makes perfect sense to me; basically, the less the mobo has to do with the actual signal processing, the better!
Hmm, thats pretty scary, I've got a beast of an amp (500w RMS into 8ohms, 900 into 4ohms). And my speakers aren't cheap or easy to move (40kg+ each) so blowing a tweeter isn't an option for me. You used physical attenuation?
Can't you just turn it right down to start with and carefully increase volume until you're at a safe level?
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Can't you just turn it right down to start with and carefully increase volume until you're at a safe level?
You would think so. ::)
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The other advantage is that, if you pick an external device with both a DAC and an ADC in it (like the TASCAM), you can use it to handle both sides of the measurement process (and switch back to your existing DAC when you're done measuring).
I use the Tascam for my mic input (ADC only), but my regular DAC for output. However, in Audiolense I am only picking one input and output device. Do you know how I do it? ;)
Can't you just turn it right down to start with and carefully increase volume until you're at a safe level?
That is what I do. I haven't had any issues with the Tascam. Also, many measurement programs let you attenuate digitally, too.
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I use the Tascam for my mic input (ADC only), but my regular DAC for output. However, in Audiolense I am only picking one input and output device. Do you know how I do it? ;)
I bet I've got an idea :)
Out of curiosity, any specific reason you don't just use the mic inputs on the UR824 directly? Is it just convenience (i.e. cable lengths, UR824 in an out of the way rack, etc.) or do you just like the ADC in the TASCAM more?
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I bet I've got an idea :)
Out of curiosity, any specific reason you don't just use the mic inputs on the UR824 directly? Is it just convenience (i.e. cable lengths, UR824 in an out of the way rack, etc.) or do you just like the ADC in the TASCAM more?
Last April I purchased a two channel AES DAC and an 8 channel AES DAC from a local designer (Ross Martin Audio). I also bought the Lynx AES16e. I wanted to try something else out and have more channels available in case I decided to go active with my mains. With this system I needed a preamp for my mic so I bought the Tascam. The Tascam has AES output so the Lynx AES16e is the ASIO input/output device in Audiolense. I wanted to try everything out and evaluate the measurement ability and playback quality before making any changes. About a month ago I finally changed all my cables from TRS/XLR to XLR/XLR and swapped gear.
So, I am no longer using the UR824 in my system. The UR824 did have a higher SNR in the measurement according to Audiolense than the Tascam and is much easier to use for measurements. Probably the only thing "better" about the AES DACs is the sound quality and lower latency (for loopback). I really think the Steinberg is the best overall audio device I've ever used.
I just received an e-mail that next week I will be getting a demo Solid State Logic Alpha-Link MX and MadiXtreme card. This has 16 channels of output for $1699 and you can later expand to 32 channels.
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Anyone used the Motu UltraLite-mk3 Hybrid? It has Mic ins and 10 x balanced analog out channels & is reasonably priced. Cannot find any technical specs on the unit.
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I can't find any specs either. The manual doesn't even list specs.
The outputs are split to a group of 8 and stereo outputs. The stereo outputs are the mains outputs according to the manual which means they are probably ASIO channels 0 and 1. The first two channels of the other 8 outputs could be duplicates of the mains, but I couldn't find any info in the manual. Also, there could be channel gaps between the mains and the other 8 channels. For example, they could be ASIO channels 4-11. This means in JRiver you would need to set the output to 16 channels and reroute to the ASIO channels you are using.
With the current 15% off at Musicians Friend it is $467 which is a nice price for 8-10 channels of D/A conversion.
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Anyone used the Motu UltraLite-mk3 Hybrid? It has Mic ins and 10 x balanced analog out channels & is reasonably priced. Cannot find any technical specs on the unit.
It looks like someone got some very basic tech specs from Motu over on the support board (see third post): http://www.motunation.com/forum/viewtopic.php?f=14&t=49197
That's not a particularly full tech description, but it hits some of the bases.
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Which all got me thinking: would there be any interest around here in something similar to Mitchco's article, except focused on crossover and speaker optimization that can be accomplished using only JRiver and freeware? Obviously it won't get you quite as far, but can get you a good distance. I say that because I'm getting ready to start work on a pair of bi-amped bookshelf speakers that I plan to finish putting together and tuning over the next month or two. Because I'll be going through the steps anyway, I could document the measurements/process as I go.
100%, yes!! ;D
-Jim
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100%, yes!! ;D
-Jim
My current plan is to try and put something together over the holidays when I have a little time off work. I'll probably do something general on measurement and speaker/room correction using JRiver and freeware first, and then do something separate on bi-amping and filter design later (because that's more technical and probably of interest to fewer people).
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My current plan is to try and put something together over the holidays when I have a little time off work. I'll probably do something general on measurement and speaker/room correction using JRiver and freeware first, and then do something separate on bi-amping and filter design later (because that's more technical and probably of interest to fewer people).
Awesome - sounds great! I have a strong interest in both subjects.
Thanks for taking the time to do this.
-Jim
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Motu Specs
Hi
I have a Motu828mk3hybrid. There are as well no specs avaialble on the manual or their website.
Ask support for it, I got some specs asking directly. However, its not as complete what I got as the RME specs for example ( I own a UFX), where the specs are in the manual and very thorough.
I am wondering why Motu does not want to publish??
Peter
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Hi, very interesting topic. I m gonna try Acourate, but could anybody tell me if my Behringer C1u condenser cardioid mic is suitable for measurement? I know, its not a dedicated measurement mic, but it is still not cheap, and has flat fr.... thanks :)
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Hi, very interesting topic. I m gonna try Acourate, but could anybody tell me if my Behringer C1u condenser cardioid mic is suitable for measurement? I know, its not a dedicated measurement mic, but it is still not cheap, and has flat fr.... thanks :)
If you have a calibration for it, it should work fine provided you make sure to point it directly at the speaker when measuring (true cardioid response falls off like crazy off-axis). If it's not calibrated, you're likely to get some anomalies, and there's no way to predict how large or small they might be. I've seen reasonably expensive microphones with truly impressive irregularities in the response (even when the "calibration" graph from the technical specs was nice and flat).
If you're going to invest in something like Acourate, it's probably worth your time to invest in calibrating your mic (if it isn't already calibrated) or to buy a calibrated microphone (parts express sells a usable one for around $50).
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Thanks, unfortunately it isn't calibrated. I need to figure out a way to calibrate it...
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two excellent primers
http://realtraps.com/art_measuring.htm
http://www.gikacoustics.com/room-eq-wizard-tutorial/
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As a long term user of JRiver and having read Mitchco’s review using Acourate I've decided to give room correction a go. Sorry for the slight thread hijack but with regard to various audio interfaces (Tascam, Steinberg, Focusrite etc.) How important is the software supplied with the various products for routing, or are all the various channel inputs and outputs available directly from within Acourate’s “ASIO LogSweep Recorder” using ASIO control, or do you need to set up channel routing from within the chosen audio interfaces software.
Sorry for the slightly daft question but I am new to this pro audio stuff.
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Unless you have a 1:1 mapping from channel to speaker in so situations then a mixer app is nigh on essential ime. For instance I use the focusrite app for all measurements so I can use a 4 way wav to drive a 2.1 output (the 2 low passes are mixed to 1 output channel),
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Thanks that makes sense, but can you please confirm whether all the various channel inputs and outputs of say a Scarlett 2i4 are available directly from within Acourate’s “ASIO LogSweep Recorder” using ASIO control. I am just trying to envisage various possibilities and how all this works together.
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Thanks that makes sense, but can you please confirm whether all the various channel inputs and outputs of say a Scarlett 2i4 are available directly from within Acourate’s “ASIO LogSweep Recorder” using ASIO control. I am just trying to envisage various possibilities and how all this works together.
I have a focusrite saffire pro 24 & acourate sees the asio channels that are backed by physical inputs/outputs IIRC. You are probably best off asking Uli directly via the user group to get the complete answer.
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I have never needed to use the mixer app for any convolution or routing setup. It is all done in either Audiolense or JRiver for me. All inputs and outputs should be available from any ASIO software including Acourate. I've used many different multi-channel audio devices (Lynx, Steinberg, Tascam, Solid State Logic, etc.) and they all worked this way with ASIO.
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I have never needed to use the mixer app for any convolution or routing setup. It is all done in either Audiolense or JRiver for me. All inputs and outputs should be available from any ASIO software including Acourate. I've used many different multi-channel audio devices (Lynx, Steinberg, Tascam, Solid State Logic, etc.) and they all worked this way with ASIO.
I think the way you do sweeps in acourate is perhaps a bit obtuse, things are definitely easier with mixing capability
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Thanks, I'm getting a much better idea how all this hangs together, all I need to now is spend some money.
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Great thread! I just got Acourate and am waiting for my USB mic to arrive. I use a Playback Designs DAC that has its own ASIO driver (USB from PC to DAC).
I read above that Acourate can only deal with 1 ASIO driver. So it sounds like you're limited to a single device that has an A/D (for the mic) and a D/A for the output? Is this correct? In other words, I can't use my 1) USB mic that does its own A/D to provide a digital input to the PC, and 2) my external PD dac that has its own ASIO driver? I would need something like a sound card that has an A/D and a D/A all in one?
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Great thread! I just got Acourate and am waiting for my USB mic to arrive. I use a Playback Designs DAC that has its own ASIO driver (USB from PC to DAC).
I read above that Acourate can only deal with 1 ASIO driver. So it sounds like you're limited to a single device that has an A/D (for the mic) and a D/A for the output? Is this correct? In other words, I can't use my 1) USB mic that does its own A/D to provide a digital input to the PC, and 2) my external PD dac that has its own ASIO driver? I would need something like a sound card that has an A/D and a D/A all in one?
I believe it is possible, the method is briefly described earlier in the thread - http://yabb.jriver.com/interact/index.php?topic=85631.msg585663#msg585663
I think this conversation on the user group is the one that describes it more precisely - https://groups.yahoo.com/neo/groups/acourate/conversations/topics/5628
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Thanks Matt. I'll check it out!
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Wrt to measuring multiple listening positions, I did try that in other DRC software and found, at least in my setup, that taking measurements in multiple listening positions did not produce a better corrected response. The best results I got was to triangulate on the listening position/speakers and take one measurement and let the speakers natural polar response deal with various other listening positions.
Hello Mitch,
I just stepped into this discussion and I'd like to comment from my admittedly biased point of view (Dirac Research)
I do not think that a single "right" positioning of the microphone, for one measurement only, does exist.
The reasoning is that measurements are different in different positions... even a few centimeters make a difference up to the point that what we measure at one ear is different from the other one.
So it's reasonable to correct the common behaviour of the different curves from different measurement positions, unless we want to listen with one ear only in a rigidly fixed single point in space :)
Here we see the eighteen curves of the nine measurements for left and right channels in the nine points that define the listening area together with the two curves (the lighter ones) that represent the averages of the respective L & R channels:
BEFORE CORRECTION:
(http://diracdocs.com/FR1.png)
AFTER CORRECTION:
(http://diracdocs.com/FR2.png)
b.t.w. some may wonder how is it possible that the averages are the ones shown in the images, seeing that the peaks are only a few dBs high while the dips are much lower... those are logarithmic scales (70 dBs plus 70 dBs add to 73 dBs and not 140)
Ciao, Flavio
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Welcome, Flavio. Nice to have you here.
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Welcome, Flavio. Nice to have you here.
Thanks Jim :)
I'm grateful for the opportunity of posting here
Flavio
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Very interesting post Flavio, thanks.
And welcome to the forum!
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Great thread! I just got Acourate and am waiting for my USB mic to arrive. I use a Playback Designs DAC that has its own ASIO driver (USB from PC to DAC).
I read above that Acourate can only deal with 1 ASIO driver. So it sounds like you're limited to a single device that has an A/D (for the mic) and a D/A for the output? Is this correct? In other words, I can't use my 1) USB mic that does its own A/D to provide a digital input to the PC, and 2) my external PD dac that has its own ASIO driver? I would need something like a sound card that has an A/D and a D/A all in one?
Or you can use the Focusrite Scarlett to give you a single point/clock for input/output. This device uses normal MIC input connections and not USB.
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I do not think that a single "right" positioning of the microphone, for one measurement only, does exist.
The reasoning is that measurements are different in different positions... even a few centimeters make a difference up to the point that what we measure at one ear is different from the other one.
So it's reasonable to correct the common behaviour of the different curves from different measurement positions, unless we want to listen with one ear only in a rigidly fixed single point in space :)
fwiw the software mitch (and I) uses allows you to correct the speaker itself separately to the effect of the room so it can still involve multiple measurements, just not multiple seating positions. It also uses a frequency dependent windowing based technique to appropriate smooth the measured response before fitting that to your target curve. i.e. I think it is aiming to see through the position specific information so as to deliver a more robust correction. The windowing parameters are themselves tunable so you can choose to include more or less of the room response in your correction.
I don't know much about exactly how you use dirac or how it works. One thought that springs to mind is how you guard against the natural off axis response of the speaker skewing what you try to correct? are there some limits to the area you should measure so as to avoid this possibility for example?
Here we see the eighteen curves of the nine measurements for left and right channels in the nine points that define the listening area together with the two curves (the lighter ones) that represent the averages of the respective L & R channels:
are those the unsmoothed responses that are then (spatially?) averaged to get the composite view? and is the after view the theoretical after or results of measuring the outcome? what class of problems do you aim to fix above 500Hz or so?
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Hello Mitch,
I just stepped into this discussion and I'd like to comment from my admittedly biased point of view (Dirac Research)
I do not think that a single "right" positioning of the microphone, for one measurement only, does exist.
The reasoning is that measurements are different in different positions... even a few centimeters make a difference up to the point that what we measure at one ear is different from the other one.
So it's reasonable to correct the common behaviour of the different curves from different measurement positions, unless we want to listen with one ear only in a rigidly fixed single point in space :)
...
Ciao, Flavio
Aloha Flavio!
Well, I have to respectfully disagree and have measurement proof :-) I used REW, with the signal routed through JRiver/Convolution engine with the FIR filter engaged, to measure the frequency response of my right speaker at 6 different locations across a 6' x 2' grid at the listening position some 9ft away. Basically the couch area:
(http://i1217.photobucket.com/albums/dd381/mitchatola/6froverlayscovering6ftx2ftarea_zpsc908e5f7.jpg)
The most ragged curve is measured with the mic the furthest away from the right speaker, i.e. back, left on my couch. But adding the left speaker in makes up for it (i.e. averages it out).
From a time coherence perspective, looking at the step response of my right speaker, shows everything arriving at the same time measured over a 6' x 2' grid:
(http://i1217.photobucket.com/albums/dd381/mitchatola/6measuresoverlay6ftx2ftarearightspeaker_zpsf63e452e.jpg)
Again, both of these charts are measured results, not simulations.
For folks interested in what a good/bad step response looks like, a good explanation can be found here:
http://www.stereophile.com/content/measuring-loudspeakers-part-two-page-2
and
http://www.stereophile.com/content/measuring-loudspeakers-part-two-page-3
Note the reason for using a step response when looking at speakers time coherence versus an impulse response. Short answer, a step response is more evenly weighted across the frequency range, whereas an impulse response is heavily weighted by high frequencies, usually the tweeter.
Audiolense also has multi-seat measurement/correction capabilities that I tried, but again, ended up that one measurement sufficed and no audible difference, to my ears, when I compared a single measurement correction versus a multi-seat correction when I AB the correction filters in real time through JRiver. So having multi-seat correction, for me, is a non-feature.
To be fair, 500 Hz on up in my system uses constant directivity horns which are designed for even frequency response coverage across a 90 x 40 degree pattern, so this could be a contributing factor to why only one measurement suffices in my system.
However, for me, having the capability to create digital XO's with a wide variety of slopes, time aligning and linearizing individual drivers, in addition to digital room correction, are must haves. There is a major audible difference between just room correction and room correction with digital XO, driver time alignment/linearization as measured and described in my article: http://www.computeraudiophile.com/content/556-advanced-acourate-digital-xo-time-alignment-driver-linearization-walkthrough/
Best regards, Mitch
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Aloha Mitch :)
Well, I have to respectfully disagree and have measurement proof :-)
Proof of what exactly?
From a time coherence perspective, looking at the step response of my right speaker, shows everything arriving at the same time measured over a 6' x 2' grid
Yes, they are similar for the first 1ms (the direct wave, as normally expected), after that they start to be different (the room)
Doing a single measurement compensation will give the best possible result in that point, but the result is unknown in any other points.
Doing a multi-point compensation will result in a compensation that is better on average in the measured points (you add more information). In your case where you fairly explained that your speakers dispersion characteristics are such that the direct wave response does not vary much with position, and the room response is similar for all positions, one measurement can be enough as it will be representative for a larger area... but this is not normally the case.
If one or both of these are not true, as usually happens, more measurements will be needed to avoid over compensating for the behaviour in that one measured point.
Jakob Agren explained that as follows:
"Mathematically it is possible to correct a room, perfectly, in a single point. For this case, only a single measurement in that single point is needed, and also any additional measurements will indeed not contribute at all, and if used, they will ruin the result in that one point.
However, it is important to note the conditions for this to be true. A single point indeed means just that, a single point, with point meaning a spot with no width nor height.
Now this is the theory, in practice we don't sit still, we move about, even if just small distances. Also we got two ears, located at different places, on average about 23 cm between them, so our ears are clearly in distinct positions. On top of this our speakers are placed in a room, introducing a multitude of reflections. The reflected sound will arrive in different points in time, with different amplitude, to each ear, so at any moment in time the sound will not be the same in these two positions. In the attached image two different measurements are shown, with the distance between them being 30cm. only. Now, which one to choose if I can have only one?
(http://diracdocs.com/30cm_dia.jpg)
Any solution trying to address these issues is a compromise. Completely predicting the reflection patterns is impossible with a small number of measurements, so an educated guess will have to do. The information used come from measurements in and around the measurement position. If you only have a single point you are effectively guessing what the result will be at your two ears. More often than not, your guess will be incorrect because the wave pattern, especially at mid and high frequencies, is fundamentally impossible to predict.
When we take several measurements, our guesses about what's going on in between these positions, become better and better. As a rule of thumb, it becomes easier to predict the behavior in nearby positions the lower the frequency is, and vice versa. That's why a subwoofer correction can work OK in most cases even if just based on a single measurement, whereas a high-performance full-bandwidth optimization require more measurements in order to guarantee that we don't end up mistakenly making things worse at some frequencies. The real problem in room correction is to make the best possible estimate of what can be corrected and what can't. You need several measurements to do a good job at that."
I conclude by saying that you have done a great job and I imagine that the measurements in your room were better than average even before correction, not only because of the dispersion characteristics of your drivers, but also because I read that you "have added bass traps behind the speakers, thick carpet/underlay from speakers to listening position, broadband absorbers on the ceiling, and back wall to help reduce early reflections" and furthermore "that the stereo is offset in the room" to avoid that resonances double their impact.
In other words I do not see any contradiction between your measurements and what has been explained about the usefulness of multiple measurements.
Ciao, Flavio
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Aloha Mitch :)
Proof of what exactly?
Proof that multi-seat measurement/correction is a red herring.
Doing a single measurement compensation will give the best possible result in that point, but the result is unknown in any other points.
Doing a multi-point compensation will result in a compensation that is better on average in the measured points (you add more information). In your case where you fairly explained that your speakers dispersion characteristics are such that the direct wave response does not vary much with position, and the room response is similar for all positions, one measurement can be enough as it will be representative for a larger area... but this is not normally the case.
If one or both of these are not true, as usually happens, more measurements will be needed to avoid over compensating for the behaviour in that one measured point.
Jakob Agren explained that as follows:
"Mathematically it is possible to correct a room, perfectly, in a single point. For this case, only a single measurement in that single point is needed, and also any additional measurements will indeed not contribute at all, and if used, they will ruin the result in that one point.
However, it is important to note the conditions for this to be true. A single point indeed means just that, a single point, with point meaning a spot with no width nor height.
Now this is the theory, in practice we don't sit still, we move about, even if just small distances. Also we got two ears, located at different places, on average about 23 cm between them, so our ears are clearly in distinct positions. On top of this our speakers are placed in a room, introducing a multitude of reflections. The reflected sound will arrive in different points in time, with different amplitude, to each ear, so at any moment in time the sound will not be the same in these two positions. In the attached image two different measurements are shown, with the distance between them being 30cm. only. Now, which one to choose if I can have only one?
(http://www.diracdocs.com/30cm_dia.jpg)
Any solution trying to address these issues is a compromise. Completely predicting the reflection patterns is impossible with a small number of measurements, so an educated guess will have to do. The information used come from measurements in and around the measurement position. If you only have a single point you are effectively guessing what the result will be at your two ears. More often than not, your guess will be incorrect because the wave pattern, especially at mid and high frequencies, is fundamentally impossible to predict.
When we take several measurements, our guesses about what's going on in between these positions, become better and better. As a rule of thumb, it becomes easier to predict the behavior in nearby positions the lower the frequency is, and vice versa. That's why a subwoofer correction can work OK in most cases even if just based on a single measurement, whereas a high-performance full-bandwidth optimization require more measurements in order to guarantee that we don't end up mistakenly making things worse at some frequencies. The real problem in room correction is to make the best possible estimate of what can be corrected and what can't. You need several measurements to do a good job at that."
Rubbish, this myth needs to be busted. You are showing full resolution frequency response graphs. What you are not telling the good readers of JRiver is that our ears do not hear at this full visual display resolution.
The prevailing science, that anyone can look up and educate themselves on, http://lmgtfy.com/?q=ears+critical+bandwidth+1%2F6+octave is that our ears critical bandwidth, while frequency dependent, is typically 1/3 to 1/6 octaves.
One link from the list above is from the Handbook for Sound Engineers and is representative of the science on the psychoacoustics of the critical bandwidths of the human ear: http://books.google.com/books?id=S4nBNZ_EJwwC&lpg=PA49&ots=eLJGYvbWyH&dq=ears%20critical%20bandwidth%201%2F6%20octave&pg=PA49#v=onepage&q=ears%20critical%20bandwidth%201/6%20octave&f=false
If one wants to get a proper "visual" of what our ears are actually hearing, then applying 1/6 octave smoothing to a full resolution frequency response display is a much more representative view.
Here is a full resolution frequency response measurement of my right speaker:
(http://i1217.photobucket.com/albums/dd381/mitchatola/fullresolutiongraph_zps1edf4e4f.jpg)
By applying 1/6 octave smoothing to the same measurement:
(http://i1217.photobucket.com/albums/dd381/mitchatola/onesixthoctavesmoothing_zps35dfe3b3.jpg)
Is more visually representative of what our ears hear based on science. The 6 x 1/6 octave frequency response measures I made across a 6' x 2' grid at the couch area in my previous post, based on one analysis measurement/correction, is practical proof of this science.
Taking your full resolution frequency response graphs above and applying 1/6 octave smoothing will show that that the two measures, relative to how our ears hear, are virtually identical. That's why multi-seat measurement/correction is a red herring.
Either you don't know the science or you choose this way to try and differentiate your product. Either way, the discussion is over for me.
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I ignore Mitch's offenses and I will answer for this Forum's readers...
If one wants to get a proper "visual" of what our ears are actually hearing, then applying 1/6 octave smoothing to a full resolution frequency response display is a much more representative view.
Here is a picture showing 9 measurements smoothed at 1/6 octave (blue traces) and the average in all of these points (the black trace)
These measurements are 3 feet apart from the central measurement from a sofa in a standard room, distance to the speakers is about 9 feet or so.
(http://diracdocs.com/room_1_6_oct.PNG)
Now, when you can choose only one of the blue ones, which one is it?
Is this blue trace representative for all of the other ones?
I think it should be fairly obvious that for this room in these positions there is not a magical one point that can be measured that will be a good fit for all of them.
Even at a very heavy handed 1/3 octave smoothing it is not hard to find frequencies where the points are more than 10dB apart, here it is:
(http://diracdocs.com/room_1_3_oct.PNG)
Flavio
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Keep it civil.
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At the risk of sticking my neck out:
@Mitchco- I absolutely agree with your statement of the scientific consensus regarding the resolution of our ears. The hitch (from my perspective) is that my own in-room measurements, even with 1/3 or 1/6 octave smoothing applied, can vary quite a lot based on relatively small changes in position (a few feet). Your measurements show strong consistency over a wide area, but (as you note) having a constant directivity horn and a well-treated room may be contributing to that consistency.
Most speakers that I've measured in untreated rooms were prone to significant positional variations in frequency response even with 1/6 or 1/3 octave smoothing applied. I can observe (at certain frequencies) a six dB or more variation between two locations three feet apart in my living room with 1/6 octave smoothing applied. And my living room setup has some features that should "tighten" variation: I have some small room treatments in place and I have a horn-loaded HF stage that controls directivity moderately well up to around 12KHz. By contrast, when I measure the little cone speakers I have in my kitchen (with no acoustical treatment and laminate flooring), I can see nearly 10 dB variations at some frequencies with 1/6 octave smoothing.
If I had based my living-room room-correction on a single position, I would have gotten poor results; the way I know that is because I tried it and the resulting correction measured very poorly (and sounded pretty bad) on the other end of the couch. It's kind of a catch-22: a given speaker/room combination may not need multiple measurements or multiple-listening-position correction, but the only way to be sure whether your setup is one of them is to... take multiple measurements at different listening positions (as you did).
I have a great deal of respect for you (you've been very helpful to me), and I always enjoy reading your posts. I just think it's a little strong to say that positional variation in frequency response is a myth or red herring. I agree 100% that measurements at different positions may not be relevant for every room, but they've certainly been relevant for me.
P.S.- I actually had been meaning to thank you; I was shopping for horn throat adapters a few months back and found a review you wrote of one on Parts Express. I grabbed the adapter based on your recommendation and it was exactly what I needed! It's a small world in online audio ;D
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I read the argument not as *for* HF correction based on multiple measurement positions but rather as *against* HF correction full stop. Correcting based on a hypothetical average seems to reduce to applying random noise unless the intent is to shape the overall contour of the signal to a preference curve (in which case you could reasonably argue you should buy different speakers). The question then becomes what secret sauce is dirac deploying that is beyond simply correcting an average?
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.....
Correcting based on a hypothetical average seems to reduce to applying random noise unless the intent is to shape the overall contour of the signal to a preference curve (in which case you could reasonably argue you should buy different speakers). The question then becomes what secret sauce is dirac deploying that is beyond simply correcting an average?
Hello Mattkhan,
I quote you when you say that correcting based on an hypothetical average is not necessarily a good idea, and that is not what Dirac Live does.
Even if advances have been made since then you may be interested in this 2009 document (eventually from page 7 "FAITHFUL STEREO REPRODUCTION") which had been partially presented at the 123rd AES convention:
http://www.dirac.se/media/12044/on_room_correction.pdf
Also thanks to the thread OP (6233638) for his welcome :)
Ciao, Flavio
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I read the argument not as *for* HF correction based on multiple measurement positions but rather as *against* HF correction full stop. Correcting based on a hypothetical average seems to reduce to applying random noise unless the intent is to shape the overall contour of the signal to a preference curve (in which case you could reasonably argue you should buy different speakers).
Whose argument do you mean (Flavio's or Mitch's)? I didn't get that out of either argument, but I'll concede I may just have misunderstood ;D
But, to your point, assuming we're talking about room correction exclusively (vice speaker correction, where HF correction can be very useful), an average can be significantly better than "applying random noise" because the room may be exerting uniform effects at HF that can be usefully corrected, and that uniformity would be revealed by multiple measurements.
An example, I have two speakers that are toed in and centered on the center seat of my couch, which means that my left seat is about 13 or 14 feet from the right speaker and about 9 feet from the left speaker, and the right seat is about 9 feet from the right speaker and 13 or 14 feet from the left speaker (i.e. they are symmetrical distance-wise). The two speakers measure almost identically when close-miced on axis.
The HF measurements at the three seats all show FR/phase nonlinearities that aren't present when the speakers are close-miced. They agree with each other in many particulars, but disagree in some particulars. Correction of some of the nonlinearities that agree among the seats has produced better (read flatter and more on target) results in all seats, but correction of the areas of disagreement based on a single measurement did not produce good results for me.
Essentially, my room appears to introduce some HF effects that are uniform across my couch region, and some HF effects which are not uniform across the same region. It may have something to do with the fact that my room is asymmetrical and my couch has it's back against a bay window about as wide as the couch (which, even with curtains, is pretty reflective).
Bottom line: from where I sit (quite literally) a room can introduce effects at HF that are consistent across a given region of the room and which can (sometimes) be usefully corrected. An average can be one way to cook out commonalities, although just looking at the multiple measurements can often tell you what you need to know.
I do agree that HF room correction is significantly less useful/relevant than low frequency room correction, I just think it can have a place if you can find agreements across the region of interest in your room.
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I ignore Mitch's offenses and I will answer for this Forum's readers...
Here is a picture showing 9 measurements smoothed at 1/6 octave (blue traces) and the average in all of these points (the black trace)
These measurements are 3 feet apart from the central measurement from a sofa in a standard room, distance to the speakers is about 9 feet or so.
(http://diracdocs.com/room_1_6_oct.PNG)
Now, when you can choose only one of the blue ones, which one is it?
Is this blue trace representative for all of the other ones?
I think it should be fairly obvious that for this room in these positions there is not a magical one point that can be measured that will be a good fit for all of them.
Even at a very heavy handed 1/3 octave smoothing it is not hard to find frequencies where the points are more than 10dB apart, here it is:
(http://diracdocs.com/room_1_3_oct.PNG)
Flavio
Again, not comparing apples to apples. Your graphs are showing the uncorrected frequency response whereas my graphs are with the DRC correction filter in the signal path. Point being, if the correction filter is doing its job, then there should not be large variations in frequency response across a couch listening area for example. This should be especially true for Dirac since the claim is that multi-seat correction does matter. Therefore, Dirac's measured results (with the correction filter in the signal path) should be as good if not better than what I show below, given my single measurement point to generate the correction filters. So let’s see the “measured corrected" responses at multiple locations around the LP.
2nd, I don’t know what kind of smoothing Dirac uses, but it appears to be very different than REW (and other measurement software I have used). Here are my six measures (with the correction filter in the signal path) at full resolution:
(http://i1217.photobucket.com/albums/dd381/mitchatola/6measurementsfullresolution_zpse04a9695.jpg)
With 1/6 octave smoothing:
(http://i1217.photobucket.com/albums/dd381/mitchatola/6measurementsonesixthoctavesmoothing_zps602f321c.jpg)
With 1/3 octave smoothing:
(http://i1217.photobucket.com/albums/dd381/mitchatola/6measurementsonethirdoctavesmoothing_zpscfc2b375.jpg)
To compare apples to apples, let's use a vendor neutral solution. Given that REW is free, runs on Windows or Mac, and has been around for almost 10 years and earned the respect of audio community, let’s use REW to compare apples to apples, with the correction filter in the signal path. Therefore, when posting frequency response graphs (vertical range 45 to 105 dB and horizontal range 20 to 20 kHz with 1/6 octave smoothing and standard 500 ms window), the comparison will be a valid one. Further, given the measurement files (.mdat) are portable, these can be exchanged and overlays can be produced for direct comparison. Are you (and other members of the forum) willing to compare apples to apples?
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My comment was just based on the graphs presented by Flavio. In isolation, they suggest to me that an average based correction would be wrong for all seats hence not v useful. i.e. the example might argue *against* the efficacy of a single measurement but it doesn't seem to do much to argue *for* multiple measurements.
The whole discussion seems a bit of a red herring tbh anyway given that Dirac doesn't correct based on that average either & so it comes back, for me, to my earlier comment of which class of problems does dirac seek to fix? It's latency budget seems relatively small next to other solutions (perhaps not much more than a typical minimum phase IIR solution if we're talking <20ms) so how much can it do in the time available?The linked paper is an interesting one that I've read before but doesn't really say much about Dirac itself other than it's not just a "simple" minimum phase system. Obviously I don't expect that to come on here and say exactly what they do, just would be interesting to read some more detail.
Fair point re your room, do you know the source of those consistent non linearities?
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Are you (and other members of the forum) willing to compare apples to apples?
do you propose a particular set of mic positions or should I just take them across the seating area as I see fit?
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To compare apples to apples, let's use a vendor neutral solution. Given that REW is free, runs on Windows or Mac, and has been around for almost 10 years and earned the respect of audio community, let’s use REW to compare apples to apples, with the correction filter in the signal path. Therefore, when posting frequency response graphs (vertical range 45 to 105 dB and horizontal range 20 to 20 kHz with 1/6 octave smoothing and standard 500 ms window), the comparison will be a valid one. Further, given the measurement files (.mdat) are portable, these can be exchanged and overlays can be produced for direct comparison. Are you (and other members of the forum) willing to compare apples to apples?
Mitch, I'm game. I usually take my measurements in Holm, but I'll retake my measurements in REW (it might take me a few days).
I have one suggestion and one question
The suggestion: a slightly finer vertical scale. With a 60dB scale (i.e. 45 to 105dB) it's very hard to see whether a given variation is say 3dB or 5dB and that's a big difference. Would you be up for a 40dB scale instead (say 60 to 100)? With the smoothed graphs everything should still be visible for most folks, right?
EDIT: The question: I ask this knowing that I don't have a good answer, and hoping to hear your thoughts. How will comparing the corrected response give us a truly apples to apples comparison if we're all measuring different speakers in different rooms? Couldn't anyone's better result be explained by a different room or a speaker with better directivity, rather than by better correction methodology/technique? I mean if someone has a 40 by 60 foot room with the speakers 15 feet away from the listening position, they're going to probably see much less variation over a six foot area than someone in a 10 by 12 room with the speakers six feet away.
Not sure what to do about that, and I'm still up to compare notes. Maybe the way to get at that is to describe the room, the speakers, and the distances of the listening positions to the speakers/walls?
Regardless, I'm definitely game, and I'll try and get something posted over the weekend.
Fair point re your room, do you know the source of those consistent non linearities?
@ mattkhan- I don't, the main effects I saw are between 2KHz and 3KHz. My best guess is the bay window, it's literally about 6 inches behind the entire listening position, which is about a wavelength at 2.2KHz.
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At the risk of sticking my neck out:
@Mitchco- I absolutely agree with your statement of the scientific consensus regarding the resolution of our ears. The hitch (from my perspective) is that my own in-room measurements, even with 1/3 or 1/6 octave smoothing applied, can vary quite a lot based on relatively small changes in position (a few feet). Your measurements show strong consistency over a wide area, but (as you note) having a constant directivity horn and a well-treated room may be contributing to that consistency.
Most speakers that I've measured in untreated rooms were prone to significant positional variations in frequency response even with 1/6 or 1/3 octave smoothing applied. I can observe (at certain frequencies) a six dB or more variation between two locations three feet apart in my living room with 1/6 octave smoothing applied. And my living room setup has some features that should "tighten" variation: I have some small room treatments in place and I have a horn-loaded HF stage that controls directivity moderately well up to around 12KHz. By contrast, when I measure the little cone speakers I have in my kitchen (with no acoustical treatment and laminate flooring), I can see nearly 10 dB variations at some frequencies with 1/6 octave smoothing.
If I had based my living-room room-correction on a single position, I would have gotten poor results; the way I know that is because I tried it and the resulting correction measured very poorly (and sounded pretty bad) on the other end of the couch. It's kind of a catch-22: a given speaker/room combination may not need multiple measurements or multiple-listening-position correction, but the only way to be sure whether your setup is one of them is to... take multiple measurements at different listening positions (as you did).
I have a great deal of respect for you (you've been very helpful to me), and I always enjoy reading your posts. I just think it's a little strong to say that positional variation in frequency response is a myth or red herring. I agree 100% that measurements at different positions may not be relevant for every room, but they've certainly been relevant for me.
P.S.- I actually had been meaning to thank you; I was shopping for horn throat adapters a few months back and found a review you wrote of one on Parts Express. I grabbed the adapter based on your recommendation and it was exactly what I needed! It's a small world in online audio ;D
mwillems, I appreciate your view and the respect is mutual. The Eminence horn throat adapter is the best I have come across and glad my review was helpful.
You will have to pardon me if I am a bit tough on vendors that make claims, but provide no proof or evidence of said claim. In this case, Dirac makes the claim that multi-seat correction does matter, but does not back up the claim with any "independent" acoustical measurements of the corrected response. While I use Acourate to generate the correction filters, I use REW to "independently" verify the correction is indeed doing what it is supposed to be doing. As you can see, I have posted a response to see if Flavio is willing to compare apples to apples.
My room is less than optimal, has minimal, but carefully placed room treatments, and my speakers are based on a 1958 design. It is a far cry from the studio/control room environments I have worked in which one can see some of the rooms and acoustical treatments here: http://www.computeraudiophile.com/blogs/mitchco/hear-music-way-it-was-intended-be-reproduced-%96-part-5-67/
If one can achieve +-5 dB across the listening window, then one has achieved what most control room monitoring achieves for producing the music we all listen to and is good enough. I achieved that with one measurement used for the correction (as have many others). In fact, I think Acourate's DRC is so good that I can be near or within that tolerance, even at a full resolution using REW's averaging (see button with red square) the 6 full resolution responses:
(http://i1217.photobucket.com/albums/dd381/mitchatola/fullresolutionaveraging_zps3aaf3980.png)
I believe there are more important contributors to my results than just room correction alone. Using Acourate's digital XO is an important factor for the reasons described in this paper: http://files.computeraudiophile.com/2013/1202/XOWhitePaper.pdf and opens the door to driver time alignment and driver linearization. That's why I posted the step response in an earlier post as I believe time coherence is as important as frequency response. For example, here is the step response of my Lynx Hilo Ad DA converter:
(http://i1217.photobucket.com/albums/dd381/mitchatola/lynxhilostepresponse48khzsr_zpsd9b60fc3.jpg)
Virtually text book response and used as a reference point.
Step response of my time aligned (but not linearized) and DRC right speaker (again achieved with a single measurement used for correction) at the LP:
(http://i1217.photobucket.com/albums/dd381/mitchatola/customspeakersstepreponse_zpsd739bd50.jpg)
Comparing that to the ideal step response in the Stereophile articles I linked to earlier, it is as good as it gets.
For comparison, here is the same speaker, but the difference is no time alignment or DRC and using 3 way passive XO:
(http://i1217.photobucket.com/albums/dd381/mitchatola/rightspeaker3-waypassivexo_zps6c51a4cf.jpg)
Massive difference, not only measurement wise, but also very audible. For me, this is a much more important factor then quibbling over a few dB from a frequency response perspective and yet another reason why I think multi-seat correction is a red herring.
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Mitch, I'm game. I usually take my measurements in Holm, but I'll retake my measurements in REW (it might take me a few days).
I have one suggestion and one question
The suggestion: a slightly finer vertical scale. With a 60dB scale (i.e. 45 to 105dB) it's very hard to see whether a given variation is say 3dB or 5dB and that's a big difference. Would you be up for a 40dB scale instead (say 60 to 100)? With the smoothed graphs everything should still be visible for most folks, right?
EDIT: The question: I ask this knowing that I don't have a good answer, and hoping to hear your thoughts. How will comparing the corrected response give us a truly apples to apples comparison if we're all measuring different speakers in different rooms? Couldn't anyone's better result be explained by a different room or a speaker with better directivity, rather than by better correction methodology/technique? I mean if someone has a 40 by 60 foot room with the speakers 15 feet away from the listening position, they're going to probably see much less variation over a six foot area than someone in a 10 by 12 room with the speakers six feet away.
Not sure what to do about that, and I'm still up to compare notes. Maybe the way to get at that is to describe the room, the speakers, and the distances of the listening positions to the speakers/walls?
Regardless, I'm definitely game, and I'll try and get something posted over the weekend.
@ mattkhan- I don't, the main effects I saw are between 2KHz and 3KHz. My best guess is the bay window, it's literally about 6 inches behind the entire listening position, which is about a wavelength at 2.2KHz.
mattkhan and mwillems, I propose that we take 6 measurements over a 6' x 2' grid to cover a typical couch area, whether sitting back in the couch or sitting upright, center, left, and right positions, if you know what I mean.
REW recommends the following for posting measurement graphs: http://www.hometheatershack.com/forums/rew-forum/934-please-read-posting-graph.html which is what I have already recommended. However, given that the file is portable, we can slice and dice at any vertical and horizontal scales as required.
mwillems, wrt your question on various room sizes/absorption coefficients, speakers directivity indexes, etc, given that we are all correcting to a similar target frequency response, the results actually should be very close. After all, I would argue that this is the raison d'être behind DRC in the first place :-) And would be a good measure of how effective a particular DRC software is working. Perhaps we can agree on a target response of say flat to 1 kHz, and using 1 kHz as a hinge point, a straight line to -6 dB at 20 kHz. Or something close enough to this as what we really want to compare is the variability across the listening area that tells us how well the DRC software is working. Make sense?
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I think I've got it; thanks for the detailed answer (both of them).
I'll try and get my ducks lined up
I believe there are more important contributors to my results than just room correction alone. Using Acourate's digital XO is an important factor for the reasons described in this paper: http://files.computeraudiophile.com/2013/1202/XOWhitePaper.pdf and opens the door to driver time alignment and driver linearization.
On that you are preaching to the choir. JRiver is my crossover (using rephase to linearize the crossover), and I've definitely seen the benefits of digital XO's and time alignment both in my step response and in improved distortion performance. Dealing with the crossover made a much larger difference for me (both in terms of improved measurement and perceived audio quality) than room correction.
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Things to consider:
- Add speaker brand to the post
- Make one close-range measurement of a speaker to get a response that above 300hz is "somewhat" anechoic. This will make it possible for us to evaluate the speaker-room interaction. Or alternatively choose a very short window and post the graph for that.
Best of luck! I will enjoy it from the sideline.
Best regards,
Mikkel
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Again, not comparing apples to apples. Your graphs are showing the uncorrected frequency response whereas my graphs are with the DRC correction filter in the signal path. Point being, if the correction filter is doing its job, then there should not be large variations in frequency response across a couch listening area for example. This should be especially true for Dirac since the claim is that multi-seat correction does matter. Therefore, Dirac's measured results (with the correction filter in the signal path) should be as good if not better than what I show below, given my single measurement point to generate the correction filters. So let’s see the “measured corrected" responses at multiple locations around the LP.
I submit to you a thought experiment:
Assume you have a single perfect speaker with a single input in a perfect room such that in two points in a couch it measures flat from DC to Nyquist/2.
That is, the speaker is perfectly flat in the two points for all frequencies.
Now we apply a filter before the speaker, let's say the filter applies a 3dB notch at 500Hz.
What do you expect the response to be in the two measured points after applying this filter?
The answer is, both points will be perfectly flat, except for a 3dB notch at 500Hz.
Now assume the original response was flat but for a 3dB notch at 500Hz in point 1 and a 3dB peak at 500Hz in point 2.
We only measure point 1, and design a filter that is flat but for a 3dB peak at 500Hz to cancel out the notch in point 1.
After applying this filter, the response in point 1 will be flat, how will it be in point 2?
Did the filter do what it was supposed to? Maybe.
Did the filter have enough information to fix the issues in point 1 and point 2 at the same time? No.
What would have been the preferred filter when taking both points into account?
Flat if point 1 and point 2 are equally important.
Can a single filter acting on a single speaker behave different in different points? No.
If we agree on the above, then we can also agree that there is no filter acting on a single speaker that can remove variations between different points (short of the special case of the 0 filter).
All you can hope to achieve is a filter that does the best possible job in all points at the same time and accept it is a compromise.
If the room varies a lot, having too few measurement points will give you a filter that is not representative for your listening area since there was not enough information at design time.
If your room has a nice and uniform response this will not be an issue.
Flavio
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I submit to you a thought experiment:
fwiw I think this link (http://"http://www.regonaudio.com/Digital%20Filters%20Part%20I.html") is a nice, simple summary of that experiment
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Hello Flavio,
I see no point in disagreeing in what is logical. One or multiple filters cannot treat the effect of room response in a differential manner. It can only take into account what is known and then try to avoid doing excessive damage in other listening positions while correcting at the main listening position (of course, the algorithm tries to correct the other positions as well but gives primacy to the main listening position). This is of course a good thing, no doubt. Not taking data from other positions into account there is a chance of making the sound worse at the other positions. If it has any practical relevance is of course another matter (see below).
Anyway, in some situations multi-seat correction may not yield much better results than single-seat correction for the reason as I just mentioned: Mainly because the main listening position has priority in the correction procedure. The corrections in all other places than the main listening seat may thus be minimal (or they may not) because they are constrained by the effects of the correction of the main listening position
I don't doubt what you are saying but it would still be interesting to see the differences in correction between single vs multi-seat correction, since the changes may be subtle (or they may be big depending on the sound at each of the measurement positions).
Btw, I do multi-seat correction. All things being equal, it is the better option - even if doing multi-point correction within a small area similar to the area just around your head. More (good) data --> better results... up to a certain point of course.
BR Mikkel
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My contribution is here - https://drive.google.com/file/d/0BxdmSMpV-t3GRTFFa18tdWJKME0/edit?usp=sharing
I couldn't manage 6 positions as I couldn't work out a way to get the mic into the 6th position without being in the way of the speakers. Floor plan is attached, naming convention for the measurements is
<mic position>-<speaker>-<DRC status>
so 1-L-On means mic on the far left of the pic, L speaker with DRC (provided by acourate hosted by jriver) on, acourate is a single measurement system and I use position 2 for measurements.
Speakers are MK MP150 Mk2 (http://"http://mkloudspeakers.com/products-info/mp-150-mk2-flat-on-wall-loudspeaker/Product/166")
Attempts at close mic measurement can be found in this post - http://yabb.jriver.com/interact/index.php?topic=88942.msg614015#msg614015 - it's a bit tricky though as there are 3 tweeters that have different crossovers for dispersion control reasons.
My comments;
- room is completely untreated except for normal furnishing, it's a lounge home cinema system I use for music
- the amount of correction for my sub is quite large, the results without correction are therefore quite meaningless so I would ignore <100Hz
- it seems I wasn't super careful with mic orientation so you might want to ignore >15kHz too
- there is certainly evidence of suboptimal correction in the other locations <1kHz
- one might expect the system to sound rather bright without correction as there is consistently a dip in the low kHz range and then it rises up again towards 10kHz before rolling off, in contrast the "on" view consistently rolls off as per the target curve
- subjectively... it sounds vastly superior with correction on, both for stereo and for surround
I've attached a graph (averages.jpg) showing the average of the 5 positions at 1/6 smoothing for the L and R with DRC on and off.
There are also some more graphs (1/6 smoothed) showing the individual collection of L and R measurements with DRC on and off.
FWIW I know the rationale of taking multiple measurements but I did take a conscious decision to optimise for a single seat (mine). Position 3 is where you go for a lie down during a film and is just outside the axis of the L speaker, attempting to account for that seems a waste of time. Position 1 is tucked into a corner and has always been rubbish so if you care about sound you don't sit there. Positions 4 and 5 would involve leaning forward so file under not going to happen. Despite all this, my wife reports that it sounds great from both those positions & much improved over no correction so what do graphs know ;D
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I like to step into this discussion and to tell a bit about my point of view:
I have seen Flavio's argumentation about multipoint measurements already at other forums. The arguments for multipoint are repeated again and again like a prayer mill. And it seems to be so very convincing. The pictures look so nice. But IMHO it is nothing else than marketing.
Flavio knows a bit more of course. We can see this by his thought experiment (http://yabb.jriver.com/interact/index.php?topic=85631.msg617344#msg617344)
So to learn more about the truth it would be nice to see only three curves instead of the 18 curves (http://yabb.jriver.com/interact/index.php?topic=85631.msg616467#msg616467) example:
1. the curve with the deepest suckout at e.g. 40 Hz (or any other frequency)
2. the curve with the least suckout at the same frequency
3. the averaged curve
Let's assume some arbitrary values. The average is 0 dB at 40 Hz (we just pick one frequency for discussion). Curve 1 shows -20 dB and curve 2 shows 3 dB.
If the target is 0 dB this means that the correction is also 0 dB as the average is already there. So listening point 1 still has -20 dB after correction, point 2 is still 3 dB.
But how is the average calculated? There are many different algorithms to calculate an average. So another algorithm may lead to an averaged curve with -6 dB at 40 Hz. To reach the 0 dB target the correction thus will boost by 6 dB. So at the listening position 1 we get -14 dB, which is an improvement and at position 2 we get 9 dB , indeed a deterioration.
So we simply learn that one correction will never fit to all positions. And indeed we apply only one correction filter.
Now let's assume we do only one measurement. And by chance we get the result with -20 dB suckout at 40 Hz. Then a target of 0 dB will lead to a boost of 20 dB (in case of primitive correction approach without boost limitation). And we already feel and know that this boost will lead to the worst correction. It may possibly sound perfect at the single nailed position but will cause a room roar elsewhere.
If we look a bit closer to the averaging of the curves in logarithmic domain we will learn that the averaged curve is more located at the upper part of all curves. It's just maths. See the 18 curves example given by Flavio. So the averaging simply spoken is nothing else than a protection against a correction overboost.
But is the averaging improving the listening result? I'm convinced that the averaging of measurements leads to an average listening quality but not top quality.
IMHO there also is an inherent flaw in the design. The example frequency responses show a steady state result (the time information is lost). We reach the example suckout of -20 dB at 40 Hz by playing a continuous sine wave. But music is usually not steady state. We can play the logsweep signal reversed. The measurement will result in the same frequency responses. But obviously it sounds different.
Acourate uses a much different approach. It looks at the sound waves arriving at the measurement position also under time aspects. In our 40 Hz example Acourate investigates in the sound wave arriving at the microphone and checks for time behaviour, transients and steady state. As the correction always is focusing on the direct sound and a short time around the direct sound (keyword: frequency dependent windowing) the steady state is much less important. Think about the sound wave radiated from loudspeakers and arriving at the ears. Is the soundwave arriving and passing at two positions of 30 cm distance really so much different as Flavio tries to convince (http://yabb.jriver.com/interact/index.php?topic=85631.msg616938#msg616938) us?
BTW it is possible to average multiple measurements by Acourate. You can even calculate the upper enevelope on multiple measurements and derive a correction if you like. Of course I have also studied all these methods during the development of Acourate. But to my knowledge up to now no Acourate user has ever been forced to do multipoint measurements and corrections.
My resume: multipoint measurements and averaging simply help to protect against over-correction. Nothing else. It does not necessarily lead to the optimal results, the averaging algorithm may be just arbitrarily selected. It is more important to look what happens in reality. A correction based on steady state signal analysis cannot be the right method.
I hope this helps
Uli
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Dear Uli,
Thanks for contributing to the discussion. Your points are of high quality but I don't see them contradicting the (theoretical) point made by Flavio.
The important contribution in your post is frequency-dependent windowing. This is used by (at least) Audiolense, your product and Audyssey.
Since you would normally use a bigger window for low-frequency correction (due to the larger wave lengths I presume?), variation (would be my guess) will show up because it is impossible to keep the speaker-room interaction out of the data. Since this part of the frequency band is not directional in our typical small rooms I think it makes good sense to use multiple measurements. As soon as travelling up the frequency band the direct sound is a function of the radiation pattern and hence the quality of the speaker, hence multipoint correction serves no important purpose (since time-domain correction only applies to the main listening position anyway).
I don't know how other products work, but with Audiolense one can set a tolerance as to the variation for the main listening position (I would expect Acourate to provide something similar). As you mention, multipoint measurements serve the purpose of not overcorrecting. It equally serves the purpose of trying to find the best balance between the quality of the main measurement position and the remaining measurement positions. In some cases this may possibly provide better results overall given the tolerance for correction for the main measurement position is large enough to accomodate the correction required elsewhere and small enough not to severely deteriorate the quality at the main measurement position.
As I mentioned in my other post, though, in other situations the variation in frequency response may be too big between various positions, meaning that in real life listening it makes little improvement for the total listening area - possibly a slight decrease in quality at the main listening position. But anything that matters? Only measurements (or simulations) combined with double blind listening tests can tell.
Anyway, I'm not expert in all this but know just enough to sort of understand how it all works (but only just).
My point here is though: Mitcho and Flavio pull the discussion into two corners instead of focusing on the cases in which both positions have their right.
Best regards,
Mikkel
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Mikkel,
multipoint measurements are quite common today. See Dirac, Audiolense, Lyngdorf, Audyssey.
I do not darn them, anyway they are better than the first approaches of room correction in the old days. Because they help to avoid overcorrection. That's it.
Indeed it is not complicated to code a multipoint measurement. You can erect big buildings around this theory and add weigthing methods and fuzzy logic.
But I definitely contradict to statements that single point measurements are wrong. You have of course to know how to interprete the measurement. IMO good skills in maths is not sufficient. You have to understand a bit more about the nature of things.
I do not think that a single "right" positioning of the microphone, for one measurement only, does exist.
The reasoning is that measurements are different in different positions... even a few centimeters make a difference up to the point that what we measure at one ear is different from the other one.
So it's reasonable to correct the common behaviour of the different curves from different measurement positions, unless we want to listen with one ear only in a rigidly fixed single point in space
Such a statement tells me that someone does not really understand. How much does the sound change when you listen to e.g. a conversation in your room and move your head... even more than just a few centimeters?
BTW frequency dependent windowing is just a part of the process. And of course the speaker-room interaction is part of the game. Otherwise we would just need to correct the speaker and it would play nicely everywhere.
Keep in mind: listening to music is NOT equal to viewing frequency responses ! Neither to a single response nor to a bunch of them. :)
Uli
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I'm pleased to see that Uli himself has stepped into this thread.
I think that there is agreement on what multiple measurements do when only frequency response is taken into account.
How much gain/attenuation limitation can result from averaging depends on the number of measurements.
As a result averaging this way works like a measurement based limitation on how much gain/attenuation there will be for each frequency.
As Uli says designing a filter this way is probably not a good idea to start with, or at least not the best idea.
A lot of information is discarded (the phase) so any solution hoping to correct the phase cannot throw this information away, and Dirac Live does this by addressing the problem from the time domain.
But in the time domain you still have the same issues... using one impulse response gives less information than using two.
Can two impulse responses being just 30 cm apart be all that different?
It depends on the speaker and the room.
Of course it is always possible to be conservative, but that will affect your end result because using multiple measurements gives more information that can be used to determine what can be compensated for and what should be left alone.
This is a very interesting subject and it could be discussed forever but the paper that I mentioned before explains these concepts with more available space: http://www.dirac.se/media/12044/on_room_correction.pdf
of course it does not go into details on how we do it, but after all Uli is reading our thread ;)
Among other things it explores what we can measure versus what we hear (i.e. from the middle of page 7)... we are looking for an ideally perfect music experience so the measurements should try to conform to that.
Ciao, Flavio
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My contribution is here - https://drive.google.com/file/d/0BxdmSMpV-t3GRTFFa18tdWJKME0/edit?usp=sharing
I couldn't manage 6 positions as I couldn't work out a way to get the mic into the 6th position without being in the way of the speakers. Floor plan is attached, naming convention for the measurements is
<mic position>-<speaker>-<DRC status>
so 1-L-On means mic on the far left of the pic, L speaker with DRC (provided by acourate hosted by jriver) on, acourate is a single measurement system and I use position 2 for measurements.
Speakers are MK MP150 Mk2 (http://"http://mkloudspeakers.com/products-info/mp-150-mk2-flat-on-wall-loudspeaker/Product/166")
Attempts at close mic measurement can be found in this post - http://yabb.jriver.com/interact/index.php?topic=88942.msg614015#msg614015 - it's a bit tricky though as there are 3 tweeters that have different crossovers for dispersion control reasons.
My comments;
- room is completely untreated except for normal furnishing, it's a lounge home cinema system I use for music
- the amount of correction for my sub is quite large, the results without correction are therefore quite meaningless so I would ignore <100Hz
- it seems I wasn't super careful with mic orientation so you might want to ignore >15kHz too
- there is certainly evidence of suboptimal correction in the other locations <1kHz
- one might expect the system to sound rather bright without correction as there is consistently a dip in the low kHz range and then it rises up again towards 10kHz before rolling off, in contrast the "on" view consistently rolls off as per the target curve
- subjectively... it sounds vastly superior with correction on, both for stereo and for surround
I've attached a graph (averages.jpg) showing the average of the 5 positions at 1/6 smoothing for the L and R with DRC on and off.
There are also some more graphs (1/6 smoothed) showing the individual collection of L and R measurements with DRC on and off.
FWIW I know the rationale of taking multiple measurements but I did take a conscious decision to optimise for a single seat (mine). Position 3 is where you go for a lie down during a film and is just outside the axis of the L speaker, attempting to account for that seems a waste of time. Position 1 is tucked into a corner and has always been rubbish so if you care about sound you don't sit there. Positions 4 and 5 would involve leaning forward so file under not going to happen. Despite all this, my wife reports that it sounds great from both those positions & much improved over no correction so what do graphs know ;D
Hey Matt, I downloaded your REW .mdat file and having a look at your L and R averages with DRC on:
(http://i1217.photobucket.com/albums/dd381/mitchatola/leftandrightaverages_zps7a411e45.jpg)
This is certainly within the +-5 dB window I mentioned in an earlier post and is +-3 dB from 100 Hz on up. Given your room layout constraints - must sound pretty darn good and looks to me that the DRC is indeed doing its job across the listening area with one measurement used for generating the filters.
Congrats!
Cheers, Mitch
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I have achieved superb sound in my living room with active speakers and Audiolense XO.
Everything changes when each driver can be corrected individually instead of correcting the speakers as one.
Multiple measurements are not needed. At least not with my setup. I have tried but never got any improvement.
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Hi,
I have been playing with the dirac trial. For practical reasons I had to move the speakers in a much worse position. So I was interested in what drc could do here. Red and green are the un corrected speakers. Blue and purple corrected. As you can see from the attached screenshot it certainly does its work in the bass region. Also imaging is a bit smaller but with better focus. And the results for most part are within Mitschko's +- 5db range. Although I think the results are positive in this setup I do prefer the results I got with Denis Sbragion open source drc software. Loopback with jrmc does not work on the htpc so I could nor measure Denis drc output with rew. I have not tried acourate or audiolense yet.
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After some feedback from dirac I redid my measerments with the mic directed fowards instead of to the ceiling as was advised in the dirac manual. This is because my mic was calibrated on-axis.
The result sounds very good to my ears. In the measurements you can see that the new filter flattens the high end a little better then the previous one.
I think it now sounds at least as good as with Denis Sbragion's DRC soft filter. Because DRC is free and the differences are small it's an easy choice to stick with the latter. Although I will really miss the virtual sound device and thus the ability to AirPlay spotify to the htpc with correction applied.
If you want to try Denis Sbragion's DRC, it's very easy with the GUI frontend made by Alan Jordan:
DRC Designer:http://www.alanjordan.org/DRCDesigner/HelpFrameset.html (http://www.alanjordan.org/DRCDesigner/HelpFrameset.html)
I attached measerements of the left and right speaker with dirac on and off.
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This subject is treated by many, including myself, in a somewhat simplistic way... on the other side a full discussion would involve a lot of math so I have asked Jakob to try translating those concepts into english wording and in pills:
* If you measure in a single point you cannot know what is common, and what is different due to room reflections, in different points in the listening area.
Without that information it's impossible to tell what is robustly invertible.
Using information from multiple measurements the inverse can be made as detailed as allowed by the spatial variability.
* By applying heavy smoothing to a single point it is often possible to capture the behavior in neighboring points as well, as the smoothing will remove most of the details.
(Minphase) filters based on such a model are likely to be spatially robust in an area around the measured position.
The resolution of the ear have very little to do with the performance of this filter, what matters is if the filter is valid for the behavior to be corrected in points around the measured point.
* For this to be true the filter will have to be smooth in most cases, but with some exceptions.
Assume you have a speaker with a jagged response placed in a room that does not affect the response significantly.
In this case the response will be similar in a large area, and it is possible to design a filter based on a single measurement that is spatially robust and has a high resolution.
But using a single measurement it is not possible to know what resolution can be used as there is no information on what is common and what is different.
In cases where the direct wave is not dominant averaging over space corresponds reasonably well with averaging over frequency.
* If you want to correct the impulse response (not just the amplitude response) and make sure the resulting filter do not introduce pre-ringings you again need information from more than one point.
It is imperative to find out what is varying and what is static, and that information cannot be synthesized from a single measurement.
Again, various smoothing techniques (in the time domain this time) can be used to make do with only a single measurements, and sometimes the smoothed model will be a good approximation of the behavior close to the measured point; sometimes it won't be.
How well a measurement can be made to represent a large area varies from room to room and speaker to speaker.
This is it, ciao :)
Flavio
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This subject is treated by many, including myself, in a somewhat simplistic way...
Yes, but I fear that "somewhat" is just an optimistic description.
* If you measure in a single point you cannot know what is common, and what is different due to room reflections, in different points in the listening area.
Without that information it's impossible to tell what is robustly invertible.
Using information from multiple measurements the inverse can be made as detailed as allowed by the spatial variability.
Different averaging algorithms will give different results of "common". Your result may be robust under mathematical aspects (e.g. invertibility). This is the same with other "common" results (calculated by power response, Sammon maps, envelopes, arithmetic mean, geometric mean, moving average, weighted average, root mean square etc.). Are the "common" results now identical or not? If not, why are they called "common"?
But does this give a real answer to the example below?
Let's assume some arbitrary values. The average is 0 dB at 40 Hz (we just pick one frequency for discussion). Curve 1 shows -20 dB and curve 2 shows 3 dB.
If the target is 0 dB this means that the correction is also 0 dB as the average is already there. So listening point 1 still has -20 dB after correction, point 2 is still 3 dB.
But how is the average calculated? There are many different algorithms to calculate an average. So another algorithm may lead to an averaged curve with -6 dB at 40 Hz. To reach the 0 dB target the correction thus will boost by 6 dB. So at the listening position 1 we get -14 dB, which is an improvement and at position 2 we get 9 dB , indeed a deterioration.
At the end we get one correction filter. IMO it should be optimal for the desired listening position. Optimal for the listener, not for the mathematician (who should clearly know, that the result will be suboptimal at other positions).
Flavio, are you really convinced by listening that single point measurements lead to wrong results? Tell me why. Or are you just repeating Jakob's arguments under marketing aspects? Please tell me what's wrong with Acourate correction filters.
Cheers, Uli
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Jakob:
I hope this discussion have been informative to the readers of this forum, and it seems it should probably be ended about now. Here are a few closing comments from my side:
1. We are not discussing any particular products here, but the merits of single vs. multi point based compensation.
2. We don't know how Acourate produces the filters it does, and we have no knowledge about any flaws the Acourate filters may have.
3. As we have stated numerous times, measuring in a single point can work just fine, but in general more robust results can be achieved by using more information. If the extra measurements contain no additional information, a good compensation algorithm will disregard them.
4. The result should be optimal for the listener, but short of hand tuning every single room together with the intended listener there need to be an algorithm to produce a filter that results in a listening experience that is better than the uncompensated system. This algorithm must have a mathematical representation as it is performed by a computer.
5. Optimality by definition implies the existence of a criterion that should hold. This criterion need to be evaluated by said algorithm. There are probably more such criterions than there are compensation techniques. And again, this is expressed using mathematics.
Flavio:
I have listened at length to another good mixed-phase solution based on one point measurement (and apparently the same principles) which I do not mention as well as I have never mentioned Acourate.
In my opinion multipoint measurements led to better results especially because I do not listen from a single fixed position... but I do not think that my judgement (as well as Mitch's one) are especially meaningful, I believe that everybody should judge with his own ears (so it is important that a two weeks free trial of Dirac Live is available)
This is all, we leave you the last word :)
thanks for your time, Flavio
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Multi point works well for DIRAC.
Single point works very well for Acourate in my room.
I can't say which measurement method is better. However, I can say that I have a better sounding sytem using Acourate. I now own both softwares. I have spent substantial time working with DIRAC. I use very good test gear.
I like the result with Acourate better. Acourate is not a fully automated process like DIRAC. It requires some learning to use it. I am still learning so I am no expert. I look forward to learning more about crossovers. :D
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Dallas,
So you've abandoned Dirac..interesting.. :)
Was it the crossover functionality that made you switch..?
Can u describe in what way acourate sounds better..?
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There's nothing wrong with DIRAC. I confess to having a wandering ear. I previously tried acourate out last year. I didn't have appropriate test/playback gear at that time. I was very impressed my second time with it. Now, my filter is even better with Uli's help.
Listening to Acourate after DIRAC reminds me of my experience listening to DIRAC for the first time after no DSP. Everything is more clear. Better resolution bottom to top. Just me experience. It's free to try.
The primary reason for my curiosity is crossover though. I have to get a Hilo first. After that, I'll work on that next. For now, I am very happy with Acourate.
Michael.
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Interesting!
Hope you start a thread at "the other" forum about Acourate and describe your findings.
Now I have licenses for both Dirac and Audiolense..maybe I should get license for Acourate too.. ::)
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I used to have license for Audiolense about 6 years ago. I was too impatient to really stick with it. So Bernt let me sell it for $100. Since then I've grown a small part of brain to allow for some DSP knowledge. ;D
I'll post more once I finish the crossovers and speaker correction filters.
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Hi,
First of all, sorry for my English... I'm new in room correction but very interested to test some software (at least, Acourate and Audiolense).
I use a Macbook Pro with JRiver MC and an Ayre QB-9 asynchronous USB DAC to play music. So, my first idea was to install Acourate on my Macbook Pro using Parallels and Asio4All and to do the measurement with a USB mic (like the UMIK-1 for example) and my QB-9 USB DAC. But after reading this topic, in order to perform good time domain corrections, I think it's not a good idea :'( Do you confirm?
I understood that I have to use the same device for input/output and avoid to use Asio4All. Is it correct?
So, is it possible to use my Macbook with Parallels without Asio4All? If yes, what equipment do you advise me to buy (mic, etc) in order to do accurate measurements and, preferably, use my USB DAC to output the sound to my amplifier?
Thank you for your help!!!