Room EQ Wizard (free measurement and parametric EQ setup software) [Text View] - Page 2

JohnPM

11-26-05, 04:21 PM

Can you tell me how to take two different sweeps and create an average, or composiste curve? I want to take measurements from two different listening positions, average them, and then correct that curve with my BFD. If this isn't possible, is there any way you can add this feature in an update?
Can't do that in the program at the moment, but I'll add it to the features list. As a workaround you could export the measurements as text, import them to Excel, do the averaging there, export the result from Excel then re-import to the Wizard. Fiddly, but should do the trick.

braidkid

11-28-05, 01:36 PM

Is there a way to tell if the soundcard in my laptop is acceptable? It looks like I have a SigmaTel C-Major Audio card installed. I'm not sure if it's full duplex.

Also, just to verify connections. All I have on my laptop is one microphone jack and one headphone jack....do I need left and right I/O?

Thomas-W

11-28-05, 02:11 PM

You need a stereo 'line-level' input, mic input won't work.

If there's no line level input, it doesn't matter whether or not you have a duplex soundcard.

BTW, I'm in the same boat as you, so an hour ago I bought a M-Audio MobilePre USB.

braidkid

11-28-05, 02:23 PM

ok, i think i see what you mean. I found where it says microphone inputs are not acceptable.

So would the creative labs mp3 USB work?

Thomas-W

11-28-05, 02:55 PM

rajdude

11-28-05, 03:59 PM

Hi John,
First of all, today is the first time I saw your application. Looks cool ! Kudos!

I had my eyes on Behringer's DSP based RTA/EQ/PM type products for some time now but I am a little skeptical about sound quality. I suspect they may degrade the fidelity of my beloved Vinyl records. You know, the folks who love Vinyl do not want to convert their audio to digital and back again any time soon. But that is off topic I guess.

I guess I will try one out someday.

I saw this note from you…

Hi Brian,

I guess you are thinking along the lines of an RTA measurement. You will get a faster and much more accurate picture of overall in-room system performance by doing a sweep measurement. The Wizard doesn't provide an RTA display, although it could - can't see the point, though :)

Regards,

I am kinda confused, I though it is the other way round, especially in the speed. a RTA is real time ( no speed here) just "real time" "right now". A sweep takes some time.

About accuracy, I wonder how a sweep is more accurate? After all you are exciting only a small band of the spectrum at one time and measuring it. This approach may eliminate any ambient noise, but exciting the whole spectrum at one time (using pink noise)....wont that be better. How....I do not know, but the people who make RTAs generally tout the "real time" ability.

Regards,
Rajiv

krabapple

11-28-05, 04:59 PM

my laptop only has mic inputs, but my PC soundcard (M-audio 2496) has RCA-type stereo inputs as well as a digital coax input. Would either be suitable for connecting a mic/RS meter (presumably using an adapter) for use with Room EQ Wizard?

rajdude

11-28-05, 05:04 PM

John,
I am using ETF alongwith their mic and preamp and Soundblaster USB MP3+. When I get all the mixer levels right in ETF and then use the Room EQ Wizard the Wizard changes the mixer level settings, sometimes causing an overload in ETF. Is the trick to calibrate the Wizard first or how do I use the same mixer levels for both the Wizard and ETF?
Thanks,
George

Sorry for my limited knowledge, what is "ETF" :o ?

JohnPM

11-28-05, 06:31 PM

Busy evening on this thread!

Time was I would have said using a microphone input was a definite non-starter, the app used to require a stereo line-level input. However, since REQ was changed to remove the use of a loopback measurement it became possible to use a mono input, and I've just done some checks with a cheap Creative PC stalk mic and it can be used, provided the mic boost is turned on. HOWEVER the results have a 15dB dip at 100Hz that starts around 80Hz and ends around 125Hz - that is a feature of the particular mic and its construction, I'm guessing, which is a shame as elsewhere it compares very well with measurements using a RadioShack SPL meter. Of course you still need an SPL meter for calibration of the SPL reading. Incidentally, the calibration process is much easier on the latest dev version of the Wizard, which I'll release in a couple opf weeks. If you want accurate results external USB soundcards are generally very good, and many have RCA connectors which is very convenient. The RS meter makes a perfectly acceptable mic with a line level output.

Rajiv, regarding RTA's and pink noise versus sweep (or for that matter MLS) measurements, you only start to see a sensible response plot from an RTA when it has had time to carry out a LOT of averaging of the input as the nature of pink noise is its random character, whilst over a long enough period it has a flat spectrum (on an RTA, -10dB/decade on an FFT) over any short period the spectrum is distinctly lumpy. An RTA can be used to perform live tuning of filters, provided the averaging is set short enough not to introduce too much delay between your adjustment and the effect showing up in the results - it is better than sweeps or MLS in that respect, but a sweep or MLS will give a much more accurate measurement and can also show phase and decay information which is impossible to extract from RTA measurements. There is some more info on sweeps and MLS here: http://support.supermegaultragroovy.com/wiki/index.php/Log_Sweeps_vs_MLS with a link to a very good paper on various methods of frequency response measurement at the end of the article (came across that site tonight looking for a link to the Sweeps paper, looks to be for some Mac software for acoustic measurement).

ETF is an excellent acoustic measurement package written by Doug Plumb, you can find it at www.etfacoustic.com. Not free, but very good value for money nonetheless.

HTH,

JohnPM

11-28-05, 07:00 PM

As a footnote to use of the mic input: I rigged up a loopback arrangement to check the mic input characteristics on my laptop, pretty poor: -3dB at 80Hz and 15kHz and very noisy (even with mic boost turned off, which was the only way to get a half decent measurement). To be avoided.

Thomas-W

11-29-05, 01:16 AM

Hi John,

A quick question, on one Windows PC the program installed with tan colors. On another Windows PC the program installed with light blue colors?

I looked for settings but couldn't find any....

What gives?

Thanks much,
Thomas

JohnPM

11-29-05, 04:13 AM

A quick question, on one Windows PC the program installed with tan colors. On another Windows PC the program installed with light blue colors?

I looked for settings but couldn't find any....

What gives?
The app picks up the system look and feel, it doesn't impose one of its own, so must be down to how the individual PCs are configured. A word of warning: have heard of one PC that was set to a theme other than one of the Windows standards and the Sun JRE crashed on startup with a null pointer exception, seems the V5 JRE doesn't like some desktop themes.

Thomas-W

11-29-05, 11:30 AM

Interesting........

The PCs have different hardware, but both have the same desktop and the same theme, Windows Classic settings with the same solid color background, (since I'm an 'old school' kind of guy ;) )

The program runs great, I was simply curious about any potentional settings.

Thanks
Thomas

jkhome

11-30-05, 08:58 PM

If you're talking about a USB Soundblaster MP3+ External Sound Card, yes that will work if you only plan on using the RS meter.

If you eventually plan on moving up to a real test mic, you should buy something like the M-Audio MobilePre USB

Thomas, what mic do you plan on using?

I went by the Guitar Center today, just to get some mini to usb cables for the BFD, but since I also needed an external soundcard, went ahead and picked up the MobilePre also.

I already had a Behringer measurement mic, think that will be ok? (I read some web reviews where folks thought that condenser mics didn't do so well with the M-audio preamp, but that was for recording music.)

http://www.behringer.com/ECM8000/index.cfm?lang=ENG

Kal Rubinson

11-30-05, 09:26 PM

The ECM8000 will work fine with the MobilePre; I use that combination.

Kal

jkhome

11-30-05, 09:47 PM

The ECM8000 will work fine with the MobilePre; I use that combination.

Kal

Thanks Kal. Have you ever ran into a problem with excessive USB cable length?

With the help of a 4 port buss and a couple of RS 10' usb extension cables (total usb lengths of no more than 16'), I will be able to run the program from my desktop.

(My wife has the laptop, if I were to crash that....!!!) :eek:

Thomas-W

11-30-05, 10:23 PM

Thomas, what mic do you plan on using?
Like Kal I have a ECM8000. I've been doing a little research and it apprears that right down to the packaging the Nady CM100 is identical to the ECM8000, and is $10 less.

http://www.behringer.com/ECM8000/index.cfm?lang=ENG

http://img3.musiciansfriend.com/dbase/pdf/man/m_275560.pdf

Nady CM100 accessory kit (http://www.musiciansfriend.com/srs7/g=live/product/images/page=1/base_pid=275560)

jkhome

11-30-05, 10:25 PM

Sounds like I'm good to go!

Kal Rubinson

11-30-05, 10:27 PM

Thanks Kal. Have you ever ran into a problem with excessive USB cable length?

With the help of a 4 port buss and a couple of RS 10' usb extension cables (total usb lengths of no more than 16'), I will be able to run the program from my desktop.

(My wife has the laptop, if I were to crash that....!!!) :eek:

I've only used it with a short USB connected to my laptop. OTOH, I have run a SpyderTV calibrator via a USB extension for a total cable length of about14feet. That was an unpowered extension. There are powered extensions, too.

Kal

braidkid

11-30-05, 11:24 PM

JohnPM

12-01-05, 04:11 AM

John, or anyone, please help clear something up for me...
I am puzzled why for the digital RS SPL you say not to compensate for C weighting. You did say that compensating for C weighting is the same as adding compensation for the RS SPL.

I have both the digital and analog RS SPL's. When I take measurements with both SPL's using C weighting, I get the same results on paper. If I were to apply C weighting to one and not the other, I would get two different results.

I hope my question makes sense as I am thoroughly confused. Please help clear this up for me.
Hi Ryan,

This goes back to comments by Ilka on the TrueRTA for Dummies thread that the analog output of the digital meter acts as if it does not go through the C weighting network, based on some measurements of analog and digital meters. On the other hand, the published schematics of the meter indicate it does go through the weighting network. Your results indicate that the weighting network is in the line on your meter. Conflicting results, perhaps there are several versions of the meter with different behaviour. I'll add a note to the help files on that.

jtgray_10

12-01-05, 09:14 AM

Great program, fairly easy and fun to work with. Couple quick questions, I ran this program using my laptop (only headphone out and mic in) and I used the setup mic that came with my Denon 2805, calibrated the levels with my SPL meter (not the RS model or I would have used that as the input) and I seem to be getting pretty good results, at least comparable to measuring sine waves by hand and plotting them. Is there a way to check the accuracy of this setup? I'm not sure how "flat" the Denon mic is, it's entirely possible that the AVR has corrections built into it to make up for any weird mic response. Also is there a way to take multiple measurements and overlay them on the same plot? Short of exporting all the data to Excel?

Thanks,

JT

braidkid

12-01-05, 10:34 AM

Thank you John,
Are you saying the digital RS SPL acts more like a true calibrated microphone in that the RS SPL compensation is not needed when using the output of the digital RS SPL?

Hi Ryan,

This goes back to comments by Ilka on the TrueRTA for Dummies thread that the analog output of the digital meter acts as if it does not go through the C weighting network, based on some measurements of analog and digital meters. On the other hand, the published schematics of the meter indicate it does go through the weighting network. Your results indicate that the weighting network is in the line on your meter. Conflicting results, perhaps there are several versions of the meter with different behaviour. I'll add a note to the help files on that.

JohnPM

12-01-05, 03:07 PM

I ran this program using my laptop (only headphone out and mic in) and I used the setup mic that came with my Denon 2805, calibrated the levels with my SPL meter (not the RS model or I would have used that as the input) and I seem to be getting pretty good results, at least comparable to measuring sine waves by hand and plotting them. Is there a way to check the accuracy of this setup?
You can get an idea of the electrical characteristics of the mic input by looping the headphone out back to mic in and making a measurement. You will probably need to turn off the mic boost (if it is on) and use a very low measurement level. When I tried that on one of my laptops it was pretty poor, as mentioned a few posts ago. Verifying the mic is a bit tricky without another calibrated mic to use as a reference, you could try some spot verifications by using the sine wave gen and comparing the wizard's SPL figure with your SPL meter's figure, but you need to place the mic and then your meter in exactly the same location (to within a fraction of a inch) and it would be easy to be misled by a slightly different positioning. However, even if the absolute accuracy of the mic is poor, if it has a smooth response you could still use the measurements to identify and correct peaks, just avoid making overall level corrections on the evidence of the mic's readings alone.

Also is there a way to take multiple measurements and overlay them on the same plot? Short of exporting all the data to Excel?
You can make measurements in different channel slots and look at them overlaid in the Measured or Corrected graph groups, the channel labels are just that, nothing stops you making a sub measurement (say) while "Left" is selected. Just need to remember what's what when looking at them :)

JohnPM

12-01-05, 03:09 PM

braidkid

12-01-05, 10:40 PM

Thank you John, I finally got everything to work tonight!!!! Simply amazing. Thank you so much John for your contribution. What used to take me 30 minutes with pen and paper now only takes 5 seconds with the sweep!!!

One question....is the sweep really as accurate as measuring in 1Hz increments? If so, simply amazing!! :-)

rajdude

12-02-05, 11:39 AM

Just an FYI to those people who are concerned about their Microphone's calibration and linearity...

You can get your mic calibrated. :D

You will have to send your mic in, they will calibrate it against a known flat mic and send you the plot and correction file. You will get your mic back (of course ;) )

I hear it may cost around 30 bucks.

I saw one such person on TrueRTA's site (in faqs section, IIRC)

-Rajiv

PS: Now how can I be sure that the mic's preamp has a 100% linear response ?

W4ZOO

12-09-05, 09:23 AM

Any Idea when the next release is coming out.

Many Thanks

JohnPM

12-09-05, 12:11 PM

Planning to make the next release on sun 18th.

jrpavel

12-12-05, 06:40 PM

JohnPM

12-12-05, 08:08 PM

Would it be possible to export the corrections that this splendid looking tool produces as Impulse Responses for possible use with the likes of my Convolver filter / plug-in (http://convolver.sf.net)?
Yes, it would. Could derive the IR from invFFT of the calculated correction filters frequency response and export as WAV (for example). I'll add it to the features list, but it won't get done until sometime around Feb (just too much on that list already :)).

W4ZOO

12-17-05, 07:17 AM

THe boy's at AV123 will be shipping the R-DES Parametric EQ unit by the end of this month.
Whats the chances of adding a unit to the Room EQ software?
It,s a USB interface.

Ken

JohnPM

12-17-05, 06:00 PM

THe boy's at AV123 will be shipping the R-DES Parametric EQ unit by the end of this month.
Whats the chances of adding a unit to the Room EQ software?
It,s a USB interface.
I haven't looked into USB comms in Java, though I believe there has been some work done recently to provide support. I'd be happy to include support for the unit, just need to know the filter parameter ranges and characteristics and details of the comms protocol, the hardest thing is to find the time to write the code and to have access to a unit to debug the interface. I've already got two models of BFD Pro that were bought just to produce the interface to them, I'm not about to go and buy another equaliser of any flavour! Surely they have a nice set-up tool of their own?

JohnPM

12-17-05, 08:07 PM

Would it be possible to export the corrections that this splendid looking tool produces as Impulse Responses for possible use with the likes of my Convolver filter / plug-in (http://convolver.sf.net)?
This turned out to be pretty easy so I've included it in the next release (tomorrow 18th Dec). Export format is 16-bit mono signed PCM WAV with the peak normalised to digital full scale.

W4ZOO

12-17-05, 09:13 PM

Thanks for the reply John,

The software is manual input only at this stage, I know future generation of software will include this, but that could be a year away.

I'm going to ask Mark L. Schifter (CEO) Nice guy and always on a 1 to 1 with all of his clients, to look at REQW and see if he would support this endeavor.

Thanks again.

Ken

I haven't looked into USB comms in Java, though I believe there has been some work done recently to provide support. I'd be happy to include support for the unit, just need to know the filter parameter ranges and characteristics and details of the comms protocol, the hardest thing is to find the time to write the code and to have access to a unit to debug the interface. I've already got two models of BFD Pro that were bought just to produce the interface to them, I'm not about to go and buy another equaliser of any flavour! Surely they have a nice set-up tool of their own?

JohnPM

12-18-05, 10:47 AM

V3.26 is now available for download from the website (linked in my signature). A lot of additions in this release, for the full list see the Revision History (http://homepage.ntlworld.com/john.mulcahy/roomeq/changehistory.html) but here are a few images to whet your appetite:

http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/images/waterfall.jpg

http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/images/csdfilled.jpg

http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/images/csdunfilled.jpg

http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/images/scope.jpg

The new waterfall displays go up to 750Hz, and those with lots of ETF measurements will be pleased to know that you can export impulse responses from ETF as .pcm files (only allows this for full range measurements), import them to the wizard and generate waterfalls from them.

The previous version is also available in case of some problem with this build. Lots of changes, so plenty of scope, but has behaved itself for me. Have fun trying it out.

bpape

12-18-05, 11:00 AM

Damn. Now I've got to get a rag and wipe off my keyboard. I'm drooling over the new enhancements. Nice work John.

J. L.

12-18-05, 11:47 AM

V3.26 is now available for download from the website (linked in my signature).John,
Either the previous version of wizardinstall.exe is being cached somewhere, or the link on your home page for the REQW still points to the 3.25 version.

My downloaded exe file as of a few minutes ago is exactly the same size as the 3.25 version and when executed states it will install the 3.25 version.

Thanks so much for this work. I'm just a beginner at learning how to set it up. I have a Delta 410 sound card in the HTPC and it is one that has its own control panel.

I was not able to get sufficient input level unless I set the (analog) RS Level meter to the 60dB scale. I'm sure it is something I don't have set up correctly so I need to re-read your help files and try it again. The Delta 410 sound card might be looking for "pro" input levels since it has the ability to be used in that capacity. In any case, it is MUCH quicker than plotting the response by hand. I have so much to learn...

Again... great work and many thanks.

Joe L.

JohnPM

12-18-05, 04:48 PM

John,
Either the previous version of wizardinstall.exe is being cached somewhere, or the link on your home page for the REQW still points to the 3.25 version.

My downloaded exe file as of a few minutes ago is exactly the same size as the 3.25 version and when executed states it will install the 3.25 version.
Must be some strange caching issue then, the web site wizardinstall.exe file is 2.17MB from 15:32 UK time today and installs V3.26 when I download it :confused:

JohnPM

12-18-05, 06:38 PM

This is in the help files, but who reads help ;)

The way ETF 5 specifies gate times is not directly comparable to the Wizard's window widths, to get similar results (e.g. from the waterfall plots) set the Wizard's window width to 1.4 times the ETF 5 gate time e.g. if ETF 5 gate time is 200ms, use 280ms for the Wizard window width.

J. L.

12-18-05, 08:46 PM

Must be some strange caching issue then, the web site wizardinstall.exe file is 2.17MB from 15:32 UK time today and installs V3.26 when I download it :confused:John,
I cleared my browser's cache and tried again. Both times previously this morning, it appeared as if it was downloading from the web, but it must have been coping from my browser's cache. This time the file size is larger and I'm sure it will load the newer version.

Sorry for any false alarm. I am using Firefox 1.5, so if anyone else has the same issue, they need to clear their browser's cache and then download.

Thanks again.

Joe L.

circularduck

12-19-05, 08:04 AM

jrpavel

12-19-05, 11:35 AM

This turned out to be pretty easy so I've included it in the next release (tomorrow 18th Dec). Export format is 16-bit mono signed PCM WAV with the peak normalised to digital full scale.

Excellent. Many thanks.

JohnPM

12-19-05, 12:53 PM

I set up and tried the REQ this weekend (hauled my computer and monitor downstairs) and I just had a question about speaker and sub interaction. I couldn't find anything specific in the help about this. I have noticed that when I have done manual measurments in the past, I have had very significant sub/speaker interaction. Simply running my sub and getting a response will not yield similar results when run with my speakers. As well, just running one speaker and a sub will not give me proper response either. As a workaround, I connected the right speaker out from my soundcard to a left and a right input on my receiver (I have a piggyback conncector that allowed me to do this easily). I then ran the sweeps and had to adjust the BFD filters manually (mostly based on the recommendations of the software, but I had to make some minor adjustments). I would then run sweeps again, and make adjustments until I found a response I was happy with.

Is this the normal way of doing this, or is there a better way that I missed?
That's pretty much what you have to do for optimal results, particularly when making initial decisions about placement of speakers and sub (when fortunate enough to have a choice). Bass content is often mono so you need to run left and right to get a representative situation, having both running can avoid the excitation of width modes that would be excited by running either speaker alone - a mode which is not normally excited should not be EQ'd. The situation is easiest with satellite-style systems as the main speakers often have so little LF that the sub contributes everything at the low end.

JohnPM

12-19-05, 06:07 PM

There is a bug with the numeric display when measuring values on the waterfall :o, the figure displayed is always for the Left channel's data (and is blank when there is no left channel data). I'll release a version with that fixed and a couple of enhancements in a day or two, in the meantime the workaround is to make measurements in the Left channel slot if you want to measure off the waterfall.

johnbomb

12-19-05, 11:19 PM

Hey John,

WOW...what an incredible new feature set! Once again, many thanks. If you are going to be adding a few new enhancements in the next day or two, is there any way you can add the curve averager that I had asked you about earlier? Either way, I really appreciate your skills and generosity.

John

circularduck

12-20-05, 08:03 AM

Thank you for changing the input volume setting to use the signal generator. I was having a heck of a time with my receivers internal signal. For some reason the signal is very low, so for me to generate anything of significant levels, I had to really turn my receiver up. I think now it will be a much cleaner way of doing it.

JohnPM

12-20-05, 05:33 PM

If you are going to be adding a few new enhancements in the next day or two, is there any way you can add the curve averager that I had asked you about earlier?
Not for the immediate release, which is just some graphics rendering improvements and a couple of minor bug fixes, but I have a few days free between xmas and new year and I'll be looking at averaging then.

Regards,

johnbomb

12-20-05, 06:35 PM

Thanks, man, that would be huge for me. Can't wait to see it!

John

braidkid

12-21-05, 04:43 PM

Do the new updates allow for any RT60 measurements or do I need a program like ETF for this? Any plans for adding RT60 measurement in the future John?

JohnPM

12-21-05, 05:43 PM

Do the new updates allow for any RT60 measurements or do I need a program like ETF for this? Any plans for adding RT60 measurement in the future John?
There are no RT60 calculations in the software at present. I'm not convinced they have anything to offer for low frequency behaviour of small rooms, compared to waterfall and spectral decay plots. There's probably some useful info to be extracted further up the range in identifying too much or too little absorption through the mid and high frequencies, so something along those lines will go in eventually - I still have a number of things to add to improve the analysis and treatment of the low frequency end, though.

jonnyozero3

12-21-05, 08:07 PM

Wait...can we measure and display spectral decay with REQ?

JohnPM

12-21-05, 08:12 PM

Wait...can we measure and display spectral decay with REQ?
Can now :) See post #291 or website.

jonnyozero3

12-21-05, 08:52 PM

Oh wow...you are my hero :) Seriously, thank you thank you many many times. That is cool. I am really looking forward to seeing how my BFD and room treatments will affect both frequency response and spectral decay down low.

Now I just need to get home after xmas and get the program working for me :p

Thanks again!

JohnPM

12-22-05, 04:24 PM

V3.27 is now available for download. Main changes in this build are:

Impulse responses are shown as absolute dB FS or % FS values rather than normalised to the impulse peak as previously (allows impulses to be sensibly compared)
Graphics rendering uses intermediate images to get rid of the sluggish cursor movement when viewing complex plots like waterfall and impulse responses
A z slider has been added to the waterfall controls to adjust the perspective effect
A button, menu entry and shortcut (ctrl+shift+J) have been added to save the graph image as a jpeg, can choose the width of the image when saving
Fixed the bug of waterfall numeric display always showing info from the left channel
Fixed a bug where a warning message that "Impulse peak is not where it should be" could appear when it should not

And finally, as a xmas treat for johnbomb :)

Added Measurement Averaging graph group allowing averaging of frequency and impulse responses (impulse response averaging should only be used for multiple measurements from the same position). For info refer to the help entry: Measurement Averaging Help (http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/help_en-GB/html/graphpanel.html#measavg)

braidkid

12-22-05, 05:18 PM

Hi John,
I just installed your newest version. I'm getting an eroneous reading when recalibrating the soundcard. I get a good sound card calibration curve, but when running the automatic measurement to check I get a funny U shaped curve. I didnt get this before when setting up the older version. Any ideas? I am using the Sound Blast MP3 USB sound card.

JohnPM

12-22-05, 07:12 PM

Have you got C weighting compensation turned on by any chance :)

braidkid

12-22-05, 07:35 PM

Yes, is that the problem? I had it turned on during the soundcard calibration. Should I turn it off and recalibrate the sound card?

JohnPM

12-22-05, 08:40 PM

Inverse C weighting is ignored during soundcard calibration as the system "knows" it is making a loopback measurement, but for any subsequent measurement it will apply C weighting compensation if it is selected. When it does that on a loopback you see the shape of the inverse C curve. I should explain in the help that C weight compensation should be turned off to make the measurement after soundcard cal that checks the cal, I'll expand on that in the next release. All looks OK with your setup, you don't need to repeat the cal.

Regards,

gravymaker

12-24-05, 05:49 PM

Now for some reason I can't do a 10hz - 200hz 1/6 octave automatic measurement any more, using the new versions?

It seems to hang on "beginning measurements"... ? Since no-one else is mentioning it, I assume it must be just me?

I can do a "SWEEP" just fine, but not any of the the octave or 1, 2, or 5hz increment measurements.

Anyone else? My JRE is up to date. Anything else I should try?

johnbomb

12-24-05, 08:43 PM

JohnPM, Merry Christmas! I didn't expect that one so soon! I'm away from my HT at the moment, spending some time with my family, but I'm chomping at the bit to get back to my laptop and update! Thanks so much, and have a relaxing and safe Christmas/New Year!

John

GinSonic

12-25-05, 11:08 AM

Hi John,

Thank You for Your marvellous work !
I have a problem with newest versions. I cannot close program anymore when started. Neither with x-button nor with file/exit - no reaction at all. JRE is 1.5.06. Last version where exit is possible is 3.25, 3.26 does not work too.

Best regards,
Dieter

gravymaker

12-25-05, 11:36 AM

Dieter - I get the same effect - if I hit "cancel" during a measurement, the "File" and other toolbar items remain greyed-out, and i have to close the app using right-click / close on the toolar

JohnPM

12-26-05, 10:53 AM

I have a problem with newest versions. I cannot close program anymore when started. Neither with x-button nor with file/exit - no reaction at all. JRE is 1.5.06. Last version where exit is possible is 3.25, 3.26 does not work too.
Hi Dieter,

This happens if you have never selected an audio input device (i.e. the app starts up showing "choose device..." in the SPL panel, workaround is to just select your soundcard as the input device then all should work OK. Will be fixed for the next release.

JohnPM

12-26-05, 11:22 AM

Now for some reason I can't do a 10hz - 200hz 1/6 octave automatic measurement any more, using the new versions?
Sorry about that Scott, will be fixed in the next release, in the meantime can use the sweep measurement (which does the job better anyway :))

gravymaker

12-27-05, 01:07 AM

Thanks John! Did I mention *thanks* for such a great program?!

Happy holidays!

GinSonic

12-27-05, 04:59 AM

This happens if you have never selected an audio input device (i.e. the app starts up showing "choose device..." in the SPL panel, workaround is to just select your soundcard as the input device then all should work OK. Will be fixed for the next release.

Hi John,

Ahhh, that's the reason ! Thank You very much for Your help :) !

ProjectorMD

12-29-05, 10:03 PM

John,

I'm just starting my quest for a properly calibrated system and I was led to your program.

Thank you very much for all the hard work you've given us.

I have a question. Has anyone used the creative labs notebook Audigy 2 ZS with Room EQ? It seems like an excellent sound card for the price and I could use it with the RS SPL without much to do. Also, would you recommend spending the extra 150 dollars for a NADY mic and phantom amplifier/preamp or just use the RS SPL as my sampling microphone?

Devan

JohnPM

12-30-05, 09:15 AM

Has anyone used the creative labs notebook Audigy 2 ZS with Room EQ? It seems like an excellent sound card for the price and I could use it with the RS SPL without much to do. Also, would you recommend spending the extra 150 dollars for a NADY mic and phantom amplifier/preamp or just use the RS SPL as my sampling microphone?
Hi Devan,

I have seen posts from people using the Audigy 2 ZS with the Wizard reporting it works OK. As for the mic, the RS meter does the job admirably, there is no need to buy a mic and preamp for low frequency work such as assessing room resonances.

Regards,

NCDave

12-30-05, 06:09 PM

I couldn't find this info in the thread.

I saw that the Creative Sound Blaster MP3+ External (a.k.a."SB Digital Music LX?") is compatible with REQW.

I kind of wanted to play with the software this weekend and the only thing I can find locally is the Creative Sound Blaster Live! 24bit External. Is this compatible with REQW?

Here (http://www.creative.com/products/product.asp?category=1&subcategory=206&product=10702&nav=4) is a link to the product page on Crative's site.

Here (http://us.creative.com/support/downloads/download2.asp?manualID=8441&Product=10702&regionID=1&Product_Name=Live%21+24%2Dbit+External) is a link to the product documentation download page (the link on this web page is to a .chm file = Compiled HTML Help File, so you'll have to save it to your computer.)

It looks like it's a bigger pain to connect it, since the channels, e.g. "front" are 1/8" stereo jacks, vs. mono-RCA jacks on the MP3+. Here's a list of I/O with descriptions from the documentation:

1. Optical SPDIF Out Connector
Connects to the Optical In connector of recording/playback devices with optical connectors (for example, MiniDisc recorders, Digital Audio Tape recorders or external amplifers).

2. Mic In jack
Connects to a microphone using a 3.50 mm (1/8-inch) mono jack.

3. Headphone jack
Connects to stereo headphones using a 3.50 mm (1/8-inch) stereo jack. Speaker output is muted when the jack is detected.

4. Center/Subwoofer Out jack (1/8")
Connects to Center/Subwoofer channel inputs on an amplified 5.1 speaker system or to the multi-channel inputs of a home theater receiver.

5. Rear Out jack (1/8")
Connects to rear or surround channel inputs of an amplified multi-channel speaker system or to the multi-channel inputs of a home theater receiver.

6. Front Out jack (1/8")
Connects to amplified stereo or multi-channel speaker systems, or to an external amplifier.

7. Line In jack (1/8")
Connects to the line output of external stereo sources (such as cassette, TV, CD player, or a MiniDisc player).

8. Coaxial SPDIF Out jack
Connects to a coaxial SPDIF cable for output to home theater systems or digital recording devices.

9. DIN jack
Connects to multi-channel speaker systems using a speaker DIN connection.

10. USB port
Connects to the computer with the USB cable.

Thank you,
Dave

bobgpsr

12-30-05, 06:33 PM

Looks OK. You would use the Front out jack (#6.) and the Line in jack (#7.) with a Radio Shack SPL meter. You calibrate by hooking one of the channels (say the right) of #6 to #7. Then you hook the RS meter to #7 and hook #6 to the sub input (or the desired AVR input --- the 5.1 analog inputs are convenient).

NCDave

12-30-05, 09:09 PM

Thank you, bobgpsr. I just picked one up tonight at CC ($50, not on sale). I should have everything I need. I checked my stash of cables and looks like I'm all set. I have the newer model RS analog SPL meter (33-4050).

Before I open the package, does anyone out there have experience with this box using REQW?

I'll post success or failure later on :)

Looks OK. You would use the Front out jack (#6.) and the Line in jack (#7.) with a Radio Shack SPL meter. You calibrate by hooking one of the channels (say the right) of #6 to #7. Then you hook the RS meter to #7 and hook #6 to the sub input (or the desired AVR input --- the 5.1 analog inputs are convenient).

Sonnie Parker

12-31-05, 03:45 AM

I've just started fiddling with the REW software... finally. I'm using the Behringer ECM8000 with amp. All connections and setup seemed to go well... calibration of mic and soundcard etc.

However, when I get to setting the target level I get this error...

http://bfdguide.ws/images/rew003.gif

Of course I continue on and it sets the level at 80db.

Then after running the auto measurement sweep I get these errors...

http://bfdguide.ws/images/rew005.gif

http://bfdguide.ws/images/rew004.gif

The sweep plays at 80db (meaures at 80db on my RS meter as well) and according to the input measurement setting it measured it at 80db when I set it up... but for whatever reasons when the automatic measurement is taken it doesn't seem to want to read/graph it high enough on the chart.

I've tried turning up the measurement output via through my receiver and via bypassing the receiver and setting the output up directly through my BFD... still get the same error.

I figure it's probably something simple I'm missing here.

Any hope is appreciated!

Sonnie Parker

12-31-05, 03:47 AM

FWIW... here's a screen shot of the setup through my receiver...

This one is at 75db while the setup directly through my BFD is at 80db and on the BFD setup the RMS Level is on up there at -28db to get to an 80db level...

http://bfdguide.ws/images/rew001.gif

http://bfdguide.ws/images/rew002.gif

Monkey_Man

12-31-05, 12:03 PM

First, nice software. I have some basic questions. I run my home theater off a HTPC using Nforce2 audio set up. I have the HTPC connected to my receiver with digital coaxial connection.

What I would like to do is use this software to graph my subwoofer. I will be getting a SVS PC-Ultra Subwoofer, which as many of you know has a built in single band Parametric Equalizer. The point would be to graph the performance and calibrate on the sub for a flat response.

Before I get the sub, I would like to learn how to use this software on my KSW-15.

1. Can I use my rat shack sound meter as the mic with the line input on it? Using the rat meter as a mic, would I still have to correct the output values to account for the meters inability to accurately measure low frequencies?

2. Any quick tips on how to use the software in this capacity?

3. As far as hook up goes, I assume all I have to do is connect the mic in the mic input on the HTPC and place the mic at the listening position. Select the proper devices for playback and recording and finally calibrate software and I'm off running.

Sorry if these questions where answered elsewhere in the tread, I do not I have the time to completely read it right now.

Thanks again!!!

Monkey_Man

12-31-05, 12:21 PM

Okay when in doubt read the help file question 1 I have answered now.

Will the software adjust for the inaccuracies of the sound meter?

JohnPM

12-31-05, 02:01 PM

Hi Sonnie,

From the soundcard loopback measurement all looks to be basically OK with the PC setup, so the problems are likely to relate to the ECM8000 and amp. With the mic/amp hooked up to the right input and the SPL meter turned on, but without any test tone playing, what rms level and SPL are shown on the meter? If you then play a pink noise sub cal signal at a level that gives about 75dB on your own SPL meter, what do the rms level and SPL figure on the Wizard's SPL meter change to?

Regards,

JohnPM

12-31-05, 02:03 PM

Will the software adjust for the inaccuracies of the sound meter?
The main source of low frequency error is due to the meter's C weighting network, if you select "Comepnsate for C weighting" in the meter menu the Wizard will apply a correction that takes account of that. Note that you need to use a line input, not a mic input.

Regards,

Sonnie Parker

12-31-05, 02:07 PM

It's definitely related to the mic and amp somehow. I switched everything over to the RS meter/mic and recalibrated everything and all works fine with it.

With no actual pink noise playing the SPL meter on the REW screen is still showing 80db. I can adjust the level on the mic amp and the SPL reading fluctuates accordingly... strange. THIS IS WITH THE ECM MIC AMP.... NOT THE RS METER/MIC.

I don't see any changes between the SPL meter with or without a signal.

I can't remember if the RMS level changes but I don't believe it does.

Sonnie Parker

12-31-05, 02:26 PM

I just switched back to the ecm mic and amp... recalibrated everything.

I noticed what I would think is a huge difference in the INPUT VOLUME. With the RS meter/mic it is around 0.06*** and with the ecm mic/amp it is at 0.491.

The RMS level db does not change regardless.

With the ecm mic/amp all connected... I can press the big RED button next to the SPL meter on the REW screen and the SPL reads 80db.

I can turn the level controls on the ecm mic amp up and down and get that SPL meter to change up and down (I think I've already said that above though).

Sonnie Parker

12-31-05, 09:14 PM

I fiddled with this thing off and on all day and could never get the Behringer to work right.

The RS meter works good so I'm just using it, BUT John, if you come up with any ideas I'd appreciate it.

JohnPM

01-01-06, 08:19 AM

I fiddled with this thing off and on all day and could never get the Behringer to work right.

The RS meter works good so I'm just using it, BUT John, if you come up with any ideas I'd appreciate it.
It sounds like you have a problem with the ECM8000 connection to the mic amp, and all you are getting out of the amp is its own noise level - changing the mic amp's gain varies that, but there seems to be no effect from sound into the ECM8000. I assume the mic amp is providing phantom power to the ECM8000? That may be a switchable feature of the amp, so make sure it is on.

Regards,

Monkey_Man

01-01-06, 01:17 PM

Does the connection from the HTPC to the receiver have to be analog? WIth the connection digital, I had to unhook the main speaker to test subwoofer. I do have separate analog output for each channel from the HTPC but I didn't go there yet. Sorry for such a newb question.

ScottH1

01-01-06, 03:12 PM

NCDave

01-01-06, 11:58 PM

This is a followup to my previous post inquiring about the compatibility of the Creative Labs Sound Blaster Live! 24 bit External sound card with the Room EQ Wizard software.

As far as I can tell, it is NOT compatible.

This does not appear to be an issue with the software. The sound card's Line In sensitivity is too low to be able to perform the sound card calibration. I double-checked all cables, connections and settings but was never able to complete this procedure. Each time, I got the "Unable to detect loopback signal" error. I tried turning the sound card's output all the way up and still got this error message.

This was not the only problem. The first one came with the "Set Input Volume" procedure. Again, with the Line In level set to maximum, the signal from the RS SPL meter (33-4050) was too low for the Line In input on the sound card.

If anyone has any ideas, I'll certainly try them out. Otherwise this dang $50 card is going back to Circuit City, where I'll no doubt have to pay a restocking fee (at best) because Creative Labs' hard blister pack packaging cannot be opened without destrying it.) :mad:

Sonnie Parker

01-02-06, 12:36 AM

It sounds like you have a problem with the ECM8000 connection to the mic amp, and all you are getting out of the amp is its own noise level - changing the mic amp's gain varies that, but there seems to be no effect from sound into the ECM8000. I assume the mic amp is providing phantom power to the ECM8000? That may be a switchable feature of the amp, so make sure it is on.

Regards,

Well that may be the problem... phantom power... there is a button for that but I didn't think I was supposed to use it. I wasn't really even sure what it was for other than I remembered back when I use to deejay that it was one button on the mixer we never wanted to touch... lol.

I'll try this and see what happens. I really know little about these mic amps anyway.

Thanks!

JohnPM

01-02-06, 07:46 AM

JohnPM

01-02-06, 07:56 AM

Hi John, I read in a previous post that you were planning to add support for the Behringer DEQ2496 to enable us to use the ECM8000 mic. Would all measurements and calibration be input via MIDI directly into your program? When do you think this feature might be added? I already have the equipment mentioned along with a DSP1124 and necessary PC, soundcard, MIDI cables etc.. but no RS sound meter or mic preamp, I would rather donate more $ towards your great program than to RatShacks bottom line. Thanks...Scott
Hi Scott,

You still need an SPL Meter to establish the actual SPL level for the Wizard as it has no absolute reference. The DEQ2496 has a different Midi implementation than the DSP1124 (which is different to the FBQ2496), the main difficulty in supporting the DEQ is having a unit to test the comms on. If I can find someone who can lend me one to debug the Midi comms I'll add the support for it.

NCDave

01-02-06, 11:29 AM

Hi Dave,

If I remember rightly on the Creative mixers the various analog inputs are all accessed via a common "analog mix" control, which is what gets selected when you choose the LINE IN input. You have to select "Analog Mix" in the Record panel of the Basic tab of the Creative Surround Mixer then go to the Source panel and mute all the sources except for Line In. The card should work OK so it is worth persisting.

Thank you, John. I think "should" is the operative word in your last sentence. I've fiddled with the mixer controls as many ways as I can find and still have the same problem. I've also tried manually setting the controls to maximize the output signal going to the Line Input.

I've attached a screenshot of the mixer controls. I've tried muting all Sources except Line In, all except Wave, all but Line In and Wave, etc. I've tried adjusting REQW's Test Input Level in the pop-up dialog box.

The soundcard has separate inputs for Mic and Line In, but only one mixer control for both. The Choose input... drop-down in REQW is set to MICROPHONE. The only other choice except "What U Hear." "Choose output..." is set to SPEAKER (the only choice). The sound card itself is set to "2.1" mode, and I'm using the front speaker output, as the instructions show for 2.1 mode. The equalizer is disabled. REQW does appear to be operating the combined Line In/Mic slider correctly. I have some confusion about which Line In/Mic slider should be operating. When I adjust REQW's input volume, the REC (red) slider on the right is operating. Is this correct? The strange thing is, if I turn the REC Line In/Mic slider all the way down and re-run the soundcard calibration, it does not fail with low level warning. But the "response" is a straight line, so that's not right either. Any setting above about 20-30% gives the low level warning again.

The front speaker output is 1/8" stereo. The Line In is 1/8" mono. I'm using a 1/8" stereo to RCA cable, picking the right channel RCA, then coupling that to an RCA to 1/8" mono cable. I've tried switching Right and Left in the hardware and in the Creative and REQW software and verified with the pink noise output that I'm doing it correctly.

The only other thing I can thing of is that I have not installed all of the Creative bloatware--only those modules which are used to control the sound card, and none of the applications. I am able to output pink noise through the card, so at least part of this is working. The Line In appears to be the trouble. (REQW was also telling me the SPL meter output was too low, although it appeared to be high enough to be able to complete that part while ignoring the warning.)

I think in the case of this application for a sound card, the fewer controls the better. I'm not entirely sure if this is user error, hardware failure (e.g. Line In), or software incompatibility, e.g. REQW selecting the correct slider controls.

NCDave

01-02-06, 02:47 PM

Well, this is a great example of how you can often solve a problem by carefully explaining it to someone else...

There is nothing in the documentation (on line, on the box, or in the manual) which says that the Line In jack is STEREO, but it is. It's not mono, as I thought. There is a PICTURE in the manual, which shows a guitar connected to the Line In (now how many guitars are stereo?), which is why I thought it was mono.

So after connecting a 1/8" stereo cable between the front out and the Line In, I now get a proper level. Whew!

Thanks...

cornercarver

01-06-06, 02:35 AM

OK, so I had a whole boatload of problems, I'm afraid. I've got an RS analog meter, digidesign mbox USB soundcard, and a mac running OS X 10.4.3. When I attempt to measure the soundcard response, no matter what I do, it always tells me there was clipping on the input, even when the input levels are turned down so low as to barely register on the meter. Certainly, the clipping light never goes on. The resulting response graph tails down to 0 (with lots of noise) after about 2Khz. Then, it also has a tendency to crash A LOT. I had to go through the setup and calibration process at least 10 times before it finally stuck around long enough to let me upload a patch to the BFD. It would crash either as soon as it finished setting the target level, as soon as it finished an auto sweep measurement, and if not at either of those points, it would usually crash after merging a 1/6 octave stepped sin measurement.

I finally produced what looked like a flat response curve, but when I used it and ran another analysis, I had a huge peak at 60Hz that wasn't there before. Window rattling huge! I went through the whole process again (including 3 crashes and restarts - after zeroing out the BFD), this time loading the the soundcard compensation file from disk, and this time, I got it to run a sweep and merge a stepped sine measurement. However, the result then claimed there were no peaks, despite a visibly obvious one. at around 60Hz. I pushed the Optimize PK gain and adjust pk gains, and when I hit find peaks, it found one this time. However, in the end, it only needed two filters to correct the problem. I called it quits for the night (5 hours after starting) and just plugged everything in.

I am pretty sure something is wrong with the interaction with my soundcard. Anyone got a clue what it might be?

I've attached the soundcard compensation file and the roomeq_wizard.log.txt, as well as my soundcard debug file.

JohnPM

01-06-06, 04:30 AM

OK, so I had a whole boatload of problems, I'm afraid. I've got an RS analog meter, digidesign mbox USB soundcard, and a mac running OS X 10.4.3. When I attempt to measure the soundcard response, no matter what I do, it always tells me there was clipping on the input, even when the input levels are turned down so low as to barely register on the meter. Certainly, the clipping light never goes on. The resulting response graph tails down to 0 (with lots of noise) after about 2Khz. Then, it also has a tendency to crash A LOT. I had to go through the setup and calibration process at least 10 times before it finally stuck around long enough to let me upload a patch to the BFD. It would crash either as soon as it finished setting the target level, as soon as it finished an auto sweep measurement, and if not at either of those points, it would usually crash after merging a 1/6 octave stepped sin measurement.
Hi Sam,

The soundcard cal file you posted is not a valid measurement, you may find better results by just not using any soundcard cal. The main problems on the Mac are that Apple provide only very limited support for the core JavaSound classes, preferring to concentrate their efforts on their own Mac-specific class library, and Apple's V5 JVM tends to crash a lot (are you using their latest version? Believe it is release 3, http://docs.info.apple.com/article.html?artnum=302412). Nonetheless people have used the Wizard successfully on the Mac. You could try relying entirely on the default audio input and output offered by the OS by simply not selecting enything in the Wizard's device selection boxes (leave them showing "choose device") and see if that helps, otherwise perhaps one of the other Mac users will chime in. Failing that, I stumbled across some commercial software for the Mac that looks good, FuzzMeasure: http://www.supermegaultragroovy.com/products/FuzzMeasure/

regards,

cornercarver

01-06-06, 12:36 PM

I'll give it a shot next time I have a few hours to play. My wife is getting sick of me tearing apart the living room every few days (lots of new components have arrive of late, including a new sub and BFD).

--sam

braidkid

01-06-06, 01:29 PM

Hi John,
Is it possible to overlay frequency response graphs so I can show several response curves on the same graph?

Sonnie Parker

01-06-06, 01:35 PM

When you start diving into the BFD Sam... be sure to check us out at the BFD Forum (http://www.nextlevelav.com/forum/forumdisplay.php?f=93).

johnbomb

01-06-06, 03:54 PM

Hey John,

Is there any way to get REQW to perform iterative curve correction? For example, I ran a sweep and had REQW find the peaks and create and optimize a filter curve, which I then sent to my BFD (man, I love this program!). However, after re-running the sweep, the measured data differed from that predicted by REQW. Can the program take the new measurements and adjust the original filters accordingly? Thanks again,

John

JohnPM

01-06-06, 05:09 PM

Hi John,
Is it possible to overlay frequency response graphs so I can show several response curves on the same graph?
Yes, one of the graph groups shows all the measurements made (up to 9) and another shows the measurements and allows any selected set of them to be averaged.

P.S. info on the various graph groups is on this help page: http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/help_en-GB/html/graphpanel.html#top

JohnPM

01-06-06, 05:14 PM

Is there any way to get REQW to perform iterative curve correction? For example, I ran a sweep and had REQW find the peaks and create and optimize a filter curve, which I then sent to my BFD (man, I love this program!). However, after re-running the sweep, the measured data differed from that predicted by REQW. Can the program take the new measurements and adjust the original filters accordingly?
Not yet, but it is on the list. If the measurements differ significantly from the predicted curve with the mic in the same place as the original measurement that can be an indication that the behaviour where the differences are is not minimum phase - generally speaking the prediction is very accurate.

Trymor

01-06-06, 07:01 PM

I can't get REW to MEASURE SOUNDCARD RESPONCE. I set the soundcard measurement level (I have tried different values), push continue and the window closes and nothing else happens. I try to SAVE SOUNDCARD MEASUREMENT AS CAL FILE, and it says "No measured data to save". But when I try to measure souncard responce again, it tells me there is already a measurement that will be removed. Through all this, nothing is displayed in the graph window (including sliders).

I then ran an automatic measurement, inversed the - and + values in the text file, loaded that file as the souncard calibration file, ran another automatic measurement, and the responce trace didn't change (like there was no calibration file loaded, eventho there was.

Any Ideas? I have a Turtle Beach Santa Crus soundcard and used a loopback cable during the previous trials.

Trymor

01-06-06, 08:01 PM

Turns out my auto update for Java was off. I just updated to the latest JRE, and now when I open REW, the window is almost blank. I have to move my mouse pointer all around the REW window, and program elements start showing up. I was able to make a soundcard calibration file and save it, but after I loaded the file, I ran an automatic measurement, and again and the responce was not adjusted flat. It looked like there was no souncard calibration file loaded at all. I can't change many settings because the REW program window never fully loads, and I cant see half of the program.

No offence, but Java programs in windows have almost always sucked. a good 75% of the ones I have tried over the years on differen PC's have had display problems, stability problems, are bloated Because they load Java runtime then leave it running, etc.... it must be easy to program in.

Guess I will have to see if I can find an older version of Java and clean up my PC and try again :(

cornercarver

01-06-06, 09:57 PM

If your PC only has java 1.4.x on it, then auto update may just update you to the latest version of java 1.4. Make sure you are runninga 1.5.x version of java.

--sam

Trymor

01-06-06, 11:55 PM

JohnPM

01-07-06, 07:57 AM

I had 1.5_4 installed, when I uninstalled that and installed 1.5_6, it totally messed up the interface of REW. I am gonna have to try and uninstall REW, Java, my video drivers and run crap cleaner, then start fresh. I am still concerned tho that REW loaded but didnt apply my soundcard calibration.
There's no need to go that far. The behaviour you described is what happens if there has been an exception in part of the GUI code, if you look in your Windows home directory (path is shown in the REW About... dialog) plain text log files are saved which will have a record of what the exception was, can just email me those. Also worth running the "Generate soundcard debug file" entry in the soundcard menu and sending that.

Regards,

Trymor

01-07-06, 12:44 PM

I uninstalled REW, Java, Display drivers (ATI Radeon 9500), cleaned the system and rebooted a few times. I installed display drivers, Java 1.5_5, installed and ran REW, and it seemd to work....till I tryed to close it. It wouldnt close untill I killed Java in the task manager. I rebooted, and again the program elements wouldnt show up in the window, and it wouldnt close.

I looked at the logs and found an acception in my .msstyle desktop. I changed it back to the Windows XP default desktop, and now the elements seem to display properly in the program window, but REW still won't close unless I kill Java in the task manager.

Looks like I will be sending you my logs....

Sorry to be a pain (I am not a computer amature either, well, mabey a Java amature ;) )

Ricketty Rabbit

01-07-06, 03:07 PM

REW still won't close unless I kill Java in the task manager.

I'm running it in an XP-Home environment with Java 1.5.0 installed. REW won't close unless I "end task" through the task manager. I haven't tried running it yet so can't comment on your other problems.

Ricketty

JohnPM

01-07-06, 04:20 PM

The "won't close" bug happens if the audio input device has never been selected, select it once and all will be well (fixed in the next release, of course :)).

Ricketty Rabbit

01-07-06, 04:37 PM

That fixed it -- thanks! Waiting for my SPL meter before I can use this.

Ricketty

Trymor

01-07-06, 04:58 PM

You beat me to it John! I was just found that out and was gonna post it.

Well, everything 'seems' to be working right, tho time will tell. I am still getting this exception though:

Exception in thread "AWT-EventQueue-0" java.lang.NullPointerException

I made my own XP install CD, and stripped out non-English languages if that helps...

JohnPM

01-07-06, 06:05 PM

I am still getting this exception though:

Exception in thread "AWT-EventQueue-0" java.lang.NullPointerException
Is this the same as in the log file you sent me before setting the audio input device? I'm away from home this weekend so don't have access to the obfuscation log to identify where the exception is happening, but I'll be back home on Monday and can look into it then.

collin

01-07-06, 09:24 PM

hi all,

have a BFD 1124 on the way and already have the RS meter and a mac laptop, but and missing the MIDI interface box. Are there any recomendations for a cheap usb midi box that will work with REW? Took a quite look at units online and am a bit confused as to what types of MIDI connections and how many channels, etc. I need to program the BFD using REW. Since I'm only going to using the midi box for this purpose, I'd like as cheap as possible.

thanks,
collin

Trymor

01-07-06, 11:08 PM

Yes John, that is one of the same exceptions as in the log I sent you. On a high note, I have taken some measurements, and all seems to be working fine. The soundcard calibration still wasn't showing on an automatic measurement, so I closed REW, reopened it, and ran an automatic plot again, and the soundcard calibration was in effect! Weird that I had to close and re-run it, but at least it works!

collin

01-08-06, 09:40 PM

have a BFD 1124 on the way and already have the RS meter and a mac laptop, but and missing the MIDI interface box. Are there any recomendations for a cheap usb midi box that will work with REW? Took a quite look at units online and am a bit confused as to what types of MIDI connections and how many channels, etc. I need to program the BFD using REW. Since I'm only going to using the midi box for this purpose, I'd like as cheap as possible.

nevermind, i picked up a MIDISPORT 1x1 USB MIDI Interface cheap on eBay. Hope it works for this.

johnbomb

01-10-06, 09:49 AM

Not yet, but it is on the list. If the measurements differ significantly from the predicted curve with the mic in the same place as the original measurement that can be an indication that the behaviour where the differences are is not minimum phase - generally speaking the prediction is very accurate.

Hey John,

Could you explain "minimum phase"? I've looked around a bit, and I don't quite get it. Also, I'm using a cardiod mic (AKG 4033) as I don't have an omnidirectional one. I haven't yet used the RS SPL meter as a mic- is it omni or cardiod? Intuitively, the pattern would affect measurement of phase. Is this correct?

Thanks,
John

JohnPM

01-10-06, 11:30 AM

Could you explain "minimum phase"? I've looked around a bit, and I don't quite get it.
The mathematical definition of minimum phase systems is probably not much use to most folk, but for our purposes there are two main characteristics of systems that would make them not minimum phase: non-linearity is the first, systems driven heavily into distortion cease to be minimum phase and their non-linear behaviour is not correctable by our linear filtering schemes. The second can be loosely termed "predictability". Correction filters for minimum phase systems can be designed that only need to know the system's behaviour up to that point in time, whereas correction filters for systems that are not minimum phase also require knowledge of the future behaviour of the system, and even with that knowledge there are sub-classes of non-minimum phase systems for which correction filters cannot be designed.

For articles on this topic and others related to room correction, written in accessible language, have a look through Robert Greene's site http://www.regonaudio.com and in particular read through the digital filters articles, http://www.regonaudio.com/Digital%20Filters%20Part%20I.html

Also, I'm using a cardiod mic (AKG 4033) as I don't have an omnidirectional one. I haven't yet used the RS SPL meter as a mic- is it omni or cardiod? Intuitively, the pattern would affect measurement of phase. Is this correct?
The RS meter is nominally omni, but at low frequencies most mics are omni. Don't quite see how the response shape would affect phase measurements.

Trymor

01-11-06, 04:55 PM

BTW John, I have been running REW now without issues it seems. Even the language exception isnt showiing up in the log anymore, for whatever reason...

Thanks.

Trymor

01-13-06, 09:39 AM

Ok, I am making many measurements trying to find the best sub position for my room. I am also taking measurements of my satellites in different positions and at different angles.

I like to look at the measurements plot to compare different positions, but there is no trace smoothing. I have to change back to the filters plot and change the smoothing from 1/3 octive, to none, and back to 1/3rd octive to get the newest trace to show up smoothed. I have tried to push the 'apply window' button, but nothing changes.

Is there a way to have each new measurement trace automatically show up smoothed? It is very time consuming (5 mouse clicks) to change back to the filtes menu for each new measurement.

Even so, I'm loving the program - first one I have tryed that gives repeatable measurements.

JohnPM

01-13-06, 10:21 AM

Having to deselect and reselect the smoothing is a bug which is fixed in the current build, will release it in a few days. Also handy to use the Alt+number shortcuts for switching between graph groups.

Trymor

01-13-06, 11:24 AM

Great news! I assume that means once the smoothing (say, 1/3rd octive) is selected in the filters group, it will be in effect for all new measurements in the measurements group.

I will do some more reading on shortcuts to make my process more streamlined. Thanks!

JohnPM

01-13-06, 11:52 AM

I assume that means once the smoothing (say, 1/3rd octive) is selected in the filters group, it will be in effect for all new measurements in the measurements group.
No, that would be too restrictive. Smoothing is a display feature rather than a measurement feature and it would not be unusual to want different smoothing (or no smoothing) on different measurements. I'd be reluctant to have smoothing active by default on a measurement as, whilst it is very useful for full range measurements, it is inappropriate for low frequency measurements aimed at modal correction.

Trymor

01-13-06, 02:54 PM

Darn. I am taking a fair amount of full range measurements to determine speaker location, and it would be much easier if a person didn't have to manually smooth each curve taken after measurement. As it is, I have to take a measurement, go to filters, apply smoothing, go back to measurements, move the speakers, take another measurement, go to filters, apply smoothing, go back to measurements, etc...

Are you saying your software should not be used for fullrange testing? I realise it's main intent is to correct room modes, but it takes fullrange measurements, and can also be used to approximate parametric filters in a Winamp plug-in called Shibatch Super Equalizer, as sort of a quick and dirty digital room correction scheme.

Perhaps an option to have display smoothing applied at graph generation?

Darren Wadsworth

01-13-06, 04:27 PM

I want to get started today usijng Room EQ Wizard.

My only concern is with EQ the mains and sub (I listen to music and care more about the sound for that than for the 5.1).

My current audio equipment we will say, consists of a computer (for music and DVD playback) connected to 2 channel Para. EQ which is connected to the reciever. Note: I know that I could just run REQWIZ right from the HTPC. However, it is not that easy to run anything other than the front end software with my current setup.

Could someone give me a detailed rundown on how to proceed (from where to connect the notebooks output).

sorry
Darren

Thomas-W

01-13-06, 05:30 PM

Could someone give me a detailed rundown on how to proceed (from where to connect the notebooks output). It won't work unless your notebook has a full-duplex soundcard, and most don't....

http://tinypic.com/f09955.jpg

Trymor

01-13-06, 05:36 PM

If your notebook does have a full duplex 'soundcard', unplug your HTPC's output from your 2 channel PEQ, and plug the output from the laptop into the EQ.

You DO NOT have to (according to the help file) connect the left line out to the left line in of the soundcard as is depicted in the last post.

John (or anyone else),

is there a good reason the left channel should be looped back? Noise, equal load on the soundcard?

Darren Wadsworth

01-13-06, 08:04 PM

Actually I do have all of the right equipment needed. I am a computer tech by trade. External sound card etc.

My main question is basically, since I am only connecting 1 channel mono to a 2.1 system (the only ones I can EQ at this time), how do I properly take the readings?

Is each channel (L, R, sub) done individually? Since this is 2.1, do I keep the sub on for testing each left and right main? Or, do I keep the sub off when doing each left and right main, then do the sub alone.
Why isn't it more accurate to send out a "stereo" sweep to all of the speakers at the same time?

Darren

mlbrand

01-13-06, 10:51 PM

Darren,

Most of us are only trying to EQ our subs, and also perhaps measure the impact the front mains are having on these frequencies. With my crossover set at 80 hz, there is definitely some "bleeding" to my main speakers of the low frequencies down to about 60 hz due to the sloping nature of crossovers. So to measure this, I used a Y-cable to feed the sound output into the left and right inputs for my main speakers. Then I EQ the subs with the output of the mains factored in. (I have fairly large floorstanding mains that go pretty low.)

I personally don't think it's necessary to try to get the low frequency curve from ALL of your 5.1 or 7.1 speakers at once, as it rarely (if ever) happens, is only affecting a very limited portion of your frequency response when it does, and your sub and mains will dominate the response curve anyway.

However, if you really want to measure the low frequency response curve from ALL of your speakers there may be a way. Hook up your outputs as stated above, and set your receiver or pre/amp to the "7channel stereo" setting (if it has that setting). I don't think PLII or PLIIx or any other setting will do it.

Good luck! ( I like your favorite quote, that's a classic!)

JohnPM

01-14-06, 07:37 AM

Hi Trymor,

The Wizard can certainly be used for full range measurements, but best to use a calibrated mic for that as the RS meter rolls off severely at the top end. I'll work something out for the smoothing :)

Regards,

JohnPM

01-14-06, 07:45 AM

The loopback connection is not used by the newer versions of the Wizard, can connect left and right outputs to the av processor if that is more convenient for getting the signal where it is needed for measurement.

JohnPM

01-14-06, 07:50 AM

Darren, I'd usually measure and correct the sub first, then measure left and right on their own to see their response and make another measurement of both left and right running together. I'd ignore low frequency peaks in either left or right that are not present in the left+right measurement as much bass is mono and the modes which disappear in the left+right measurement probably won't get excited. Then I'd make measurements of left+sub, right+sub and left+right+sub to see how the speakers integrate and look at the effects of changing the crossover frequency and the sub phase to get the smoothest overall response.

Trymor

01-14-06, 11:33 AM

Now thats a nice detailed response for Darren! Darren, you might want to look into using your HTPC as a parametric EQ, since it is the source already. There are even full blown Digital Room Correction programs/plug-ins for both Windows and Linux (although I don't know if any of them deal with pre-echo).

Trymor

01-14-06, 11:40 AM

John,

Thanks! I know the RS meter isn't accurate enough for say, speaker building (I wish it were since I already have it) but I have a calibration file with as much as 12 db correction at 20 KHz. That should make it good enough to observe general room effect on main speaker placement.

Darren Wadsworth

01-15-06, 07:38 PM

John,

Just the info is was loking for. Thank you! So, it is ok to split the mono output from the software to both left and right of the EQ input? Just want to be sure. BTW, tried out the software (with ECM8000 and Creative USB external) yesterday just to get the feel for things. At the listening position there is a tremendous null at 70hz!! I have a long way to go to figure this stuff out. Is there a guide available to help with understanding the results of the tests? Thanks for the software and the help!!

Trymor,
"you might want to look into using your HTPC as a parametric EQ, since it is the source already. There are even full blown Digital Room Correction programs/plug-ins for both Windows and Linux (although I don't know if any of them deal with pre-echo)."

I would actually like to do that. I can't find a solution that works with my playback software (Foobar2000). If you know of software that I can use to accomplish this, please let me know.

Thank you
Darren

Trymor

01-15-06, 08:29 PM

Darren,

Go here to learn about it - http://www.duffroomcorrection.com/wiki/DRC . Neccesary plug-ins for Foobar are listed there also - http://www.duffroomcorrection.com/wiki/Foobar2000 .

I use Winamp, not Foobar, but I have been reading a little on the subject, and it seems a bit complicated. For now, I have just been using an infinite band parametric EQ plugin for Winamp to deal with the sub and sub/main integration. Don't quote me on this, but I think REW allows you to save an impulse responce that can be used with DRC and Foobar.

Oh, and yes you can split the mono output to your EQ input, or just use the stereo out of the soundcard to the stereo input of the EQ like I do.

Darren Wadsworth

01-15-06, 10:50 PM

I am learning more about the convolver approach. There is a plugin for foobar. Couldn't I generate a filter file from REQW and use that for Foobar2000?

Darren

Darren Wadsworth

01-15-06, 10:51 PM

edit: sent the last post twice somehow

Trymor

01-16-06, 07:04 AM

Thats basically what I ment by 'impulse response'. I don't know if the plug-in requires that, or the filter file from REW.

Here is a 'digital room correction' thread I just started reading: http://www.avsforum.com/avs-vb/showthread.php?t=283878&highlight=digital+room+correction

Randy Mathis

01-16-06, 04:16 PM

Thank You for the program.

I had a difficult time getting sound from my soundcard. I eventually gave up and ran the spdif out from the soundcard to my receiver and everything worked great.

I input the programs filters manually and added Sonnie Parkers recommendations for additional filters at 1.0, 1.25, 1.63, and 2.0 khz.

I then ran another sweep and the program said that there are no peaks.

I have not checked the work with another source or a manual check but I am satisfied.

I have had the BFD sitting around, unused for months. I feel better now. :D

JohnPM

01-16-06, 04:27 PM

Randy, you're welcome :)

JohnPM

01-16-06, 06:39 PM

V3.28 is now available for download, changes are detailed in the Change History (http://homepage.ntlworld.com/john.mulcahy/roomeq/changehistory.html)

This version adds a "House Curve" option to load a desired target shape for the filters to be optimised against e.g. to allow a small rise in the response at LF for a subjectively better response. It is also now possible to change the channel names (Left, Right, etc) by right clicking on the channel select tabs in the Filters pane. The new names are remembered for the next startup, as are the speaker type and bass management cutoff settings. The settings last used for the Set Extents dialog are also remembered. Fractional octave smoothing has been made much faster, 1/2 octave has been added as a smoothing step and a keyboard shortcut, ctrl+3, has been added to provide a quick way to apply 1/3rd octave smoothing to the current channel. Filter BW is now shown in Hz alongside the Q/BW figure. Various other minor changes have been made, and the "would not close" bug and broken stepped sine measurement have been fixed.

Darren Wadsworth

01-16-06, 07:27 PM

John,

I am learning more about the convolver approach. Can I use your software to generate a filter file and use that for Foobar2000's convolver? Are there instructions in the help docs for generating such a 2 channel filter?

Thank you
Darren

JohnPM

01-17-06, 04:17 AM

Hi Darren,

I don't know what the foobar2000 convolver plug-in requires. The File -> Export -> Channel Filter Impulse Response as WAV option generates the impulse response of the filters for the current channel in WAV format, written as 16-bit mono PCM signed data, with the impulse peak normalised to digital full scale. If something different is required it could be generated with a WAV editor of some sort in the short term, longer term I could add an option to the app if someone can tell me what is needed.

jrpavel

01-17-06, 08:33 AM

My convolver reads and uses mono wavs fine (although it would be even better if they were floats).

It will also take stereo wavs and apply them channel by channel to stereo input (which is mainly useful for reverb effects, but can also give a good room correction sound, despite the theoretical incorrectness). I imagine that this is what foobar2000 does, but have not checked.

John

JohnPM

01-17-06, 10:16 AM

My convolver reads and uses mono wavs fine (although it would be even better if they were floats).
Floats would be a little awkward, but I'll add a selector to choose 16, 24 or 32-bit signed PCM.

Trymor

01-17-06, 11:46 AM

Thanks John!

REW now has 2 features (basically ;) ) I asked for, and 2 more I was going to ask for. Keep up the good work!

CTRL+3 really speeds up taking multiple measurements for mains placement, and the house curve is something I wanted to ask for, and it is great to have!

Trymor

01-17-06, 11:54 AM

jrpavel,

Does this mean all we need for basic room correction in Windows is REW and your convolver plug-in? How do we get REW's mono file to convolve with a stereo source? Do we have to use REW to capture each channels impulse response separately, then combine them into a stereo file in another program (i.e. Audacity)?

jrpavel

01-17-06, 02:21 PM

Yes.

Others with more experience of room correction may want to chip in, but there are several approaches of increasing complexity.

Assume that you are using stereo, then:

You can measure the impulse response of your left and right speakers separately, and apply the results to the respective channels. The first example on http://convolver.sourceforge.net/configegs.html shows the convolver config file that allows you to do that. Alternatively, you can combine the mono impulse responses into a single stereo WAV file and load that directly instead of using a config file.

That is not, however, the theoretically best way of correcting your system, although it may give good results in practice and has the merit of simplicity. (At least not if you don't listen with a board in front of you that prevents the left channel sound from reaching your right ear ...)

The more correct way, apparently, is to take 4 impulse response measurements (left speaker to left ear, left to right, right to right and right to left) and then apply them using the head-related transfer function (HRTF) network example on http://convolver.sourceforge.net/configegs.html as a model.

I'll leave it to others to comment on / correct this advice, but the long and the short of it is that with convolver you can mix input channels, filter the result, mix/sum the results of different filter paths, and direct them to specified output channels in a completely configurable way. (In fact, I am trying to decide how best to reduce the flexibility, as it makes convolver more complex to set up than is probably strictly necessary.) So, if you want to do other things like split a channel into different frequency ranges and send the results to different output channels, you can do it.

Darren Wadsworth

01-17-06, 02:43 PM

John,

It is refreshing during these times, that you are able to devote as much of your time as you do to the programming and support of this software. Even to just offer that you would consider possibly adding Foobar2000 convolver support to this software is greatly appreciated.
I personally have listened to all of the software players out there. Foobar2000 just seems to produce the best quality for my system.

You prob already have read this thread regarding DRC:

http://www.avsforum.com/avs-vb/showthread.php?t=283878&highlight=drc

There is a guide for using this DRC for creating convolver files. It is fairly complicated and involves several programs. If your software could accomplish this, it could be as simple as taking readings saving a file, import into Foobar and done!

Example Foobar2000 files
http://www.foobar2000.org/impulses.zip

convolver plugin
http://www.foobar2000.org/foo_convolve.zip

Thank you
Darren

jrpavel

01-17-06, 04:07 PM

Darren, thanks.

I had not thought of adding plugging convolver into fb2k as I think that there already is one for it.

I've had at the SDK and there is little documentation, so it would be a matter of trying to do it by looking at the sources. It's probably not v difficult but all takes time. A VST interface is also in demand, which is reasonably well documented so that is probably higher on the list.

NB. Convolver will play the impluses, but not generate them

Trymor

01-17-06, 08:43 PM

John and jrpavel,

One thing I am still not clear on - Do we export the channel impulse response as wav, or the channel filters impulse response as wav for use with convolver?

John,

Since it seems using a stereo wav for convolution is easier than 2 mono files, might it be possible to have an option to save 2 channels as a stereo wav file?

jrpavel,

Thanks for the explanation. If I have any more questions specifically about your plug-in, I will head over to your convolver thread.

Darren,

My thoughts exactly! the DRC site makes the whole process sound complicated and using REW to make the nessesary file for convolving would be much less involved.

Darren Wadsworth

01-17-06, 09:22 PM

jrpavel there is already a convolver plugin for foobar 2k. Although I do not i know yet how well it works.

I am going to try a few experiments this weekend with DRC and/or REQW. I will take to mono readings from REQW and combine them in Cool Edit. We will see what happens!!

trymor,

What do you think? (2 spkrs and sub). Scan left channel with sub. Scan right channel with sub. Combine the two together in cool edit? Something like that. Computers I know. I am a computer tech. However, I currently know nothing about convolving (is that a word?).

Darren

JohnPM

01-18-06, 04:18 AM

Darren, Trymor,

It is the channel filters impulse response you need to export for use with the convolver. I'll look at allowing a pair of responses to be exported as a stereo WAV, for the moment Cool Edit or Audacity or similar should do the trick.

GinSonic

01-18-06, 05:52 AM

@Randy Mathis:
I input the programs filters manually and added Sonnie Parkers recommendations for additional filters at 1.0, 1.25, 1.63, and 2.0 khz.
Since I could not find these recommendations in this thread, what are they for ? Also for subwoofer calibration only ?

Thanks, Dieter

Trymor

01-18-06, 07:42 AM

Darren,

Short answer, yes, I think so, but it may depend on how your receiver handles the sub. Is it connected to the sub out of the receiver, or are you running speaker wires to the sub, then to the mains utilizing the subs high-pass crossover and turning off the sub output of the receiver.

Don't know if that really matters, but I now understand jrpavel's statement about the method not being 'politically' correct. When you take separate left and right measurements, there are nulls and peaks that are NOT there when you take a combined measurement, because they get cancelled out. So, if the left and right channels get corrected separately, the combined output will be different than predicted (i.e. different nulls and peaks than when measured with no correction). So does this mean a better result can be obtained using one system measurement (mono signal sent to left, right and sub at the same time) and using the same mono wav for both left and right channels? I don't know, but I think I am going to try it. In my mind, the corrected system response I get in REW should then be what I hear. The thing is, in real music, you have some 'mono' sounds (most low bass and centered singers) and 'stereo' sounds (guitars on the left, ambiance and soundstaging cues for example) so it seems both simple methods have pluses and minuses. That must be why the correct way involves summing and differential correction.

If we are to continue this convolver conversation, perhaps we should take it to an appropriate thread?

GinSonic,

I was wondering the same thing. Supposedly it is not a good idea to use the BFD for anything other than a sub because of the sampling errors and phase shift it creates, making the sound worse.

ebr

01-18-06, 08:27 PM

Darren Wadsworth

01-18-06, 11:03 PM

Trymor,

About moving the conversation... It's more actually related to REW and weather John thinks what the best approach for this calibration for his software would be. Also whether he wants to possibly add the ability to create file suitable for convolver from within his software.
Really, IMHO, that jrpavel is prob correct (not exactly sure there is a correct way due to the "fluid" nature of stereo sound) in using averaging. I just don't quite know enough about acoustics and audio streams to make the conversation interesting. I am going to try something with this info, this weekend using REW. We'll see how it goes. I am going to compare the readings taken with R then L then R+L and see how much different they are.

Darren

WarnerL

01-18-06, 11:23 PM

JohnPM or others,

What is a quick and dirty way to figure out (using test signals and the REW) if the line output from my Radio Shack digital SPL meter is c-weighted compensated or unweighted (as apparently some are according to other posts on this forum)?

JohnPM

01-19-06, 04:48 AM

Please forgive me if this is already covered (I searched the thread and didn't come up with anything) but how does a program like this and the BFD relate to something like the auto EQ capabilities of Receivers like the Pioneer Elite 74? Same thing, or can I still do more/a better job with this combo?
Short answer is you should be able to do a much better job at the low end with a BFD and appropriate measurement software, but the process is a lot more involved than pressing a button and sitting back. Auto EQ features would typically provide correction over most of the frequency range which could be done with a BFD or other equaliser but would need a better microphone than an RS meter for frequencies above about 2kHz.

JohnPM

01-19-06, 04:50 AM

About moving the conversation... It's more actually related to REW and weather John thinks what the best approach for this calibration for his software would be. Also whether he wants to possibly add the ability to create file suitable for convolver from within his software.
I've added the ability to generate stereo WAV files with two filter impulse responses in the current dev build so that will be in the next release.

JohnPM

01-19-06, 04:52 AM

What is a quick and dirty way to figure out (using test signals and the REW) if the line output from my Radio Shack digital SPL meter is c-weighted compensated or unweighted (as apparently some are according to other posts on this forum)?
Without some other reference measurement you can't. It is more likely that the output does need C weighting compensation than not.

Trymor

01-19-06, 08:06 AM

John,

Thanks! I may not get to doing drc till the next release anyway, as I have used the software PEQ plug-in for winamp on the bass region and it sounds great now!

Darren,

I apologize. I know the conversation was about convolution and REW, but I thought it started moving toward drc theory and setup, and I didn't want to plug up Johns thread (which I guess I did with my last somewhat lengthy post).

ebr

01-19-06, 08:37 AM

Thanks, John. I've done my own room corrections using TrueRTA and a Bijoux analog EQ so I understand how involved that process is. Looks like your program helps it out quite a bit. I just wanted to know what everyone thought of the capabilities of the auto room corrections built in to the latest equipment. I heard that the new Pio Elite receivers had added some low end corrections - but I'm not sure how good they are.

I plan to try and tackle most of the room issues with treatments, but there's always some EQ that needs to be done.

Thanks again for your generous work on this program.

Darren Wadsworth

01-19-06, 01:04 PM

""I apologize. I know the conversation was about convolution and REW, but I thought it started moving toward drc theory and setup, and I didn't want to plug up Johns thread (which I guess I did with my last somewhat lengthy post).""

No need to apologize. You are prob right about plugging up the thread. Prob going off topic. I enjoyed your lengthy post. It gave me some valuable info to think about. I need to finsh reading the DRC thread. Good information over there.

John,

Thank you for deciding to add that in. As soon as that release is ready, I will be testing that out.

Darren

Darren Wadsworth

01-20-06, 04:00 PM

I used REQW last night to create convolver files for Foobar2000. It seemed to work great! I did have one dumb question though. I know that I am supposed to export channel filters impulse response as wav. My dumb question is, is this done after I adjust the filter settings within REQW?
Also, for John, why does the program only auto adjust filters at and below 500hz? Is this because of the limitations of the RS SPL? I am using a full range mic. So any major peaks after 500hz I have to look for and adjust manually. This is very time consuming. Is there anyway that you could add the ability to adjust filters for up to 20k hz?

I tried using the DRC method also. I am a computer tech so the proceedures were not that hard for me. However, the filters it created were not correct. The music was garbage.

Thank you
Darren

Trymor

01-20-06, 07:53 PM

Wow, thats great that REW does drc files better than DRC. Yes, export the filters as wav after all your adjustments are done, or you wont be correcting anything.

I was wondering the same thing about the 500Hz cutoff, but I'm not sure if the precison can be good enough considering the automatic filters are far from giving me flat response. I have to add a bunch of filters, and tweak the supposedly optimized ones, but at least it gives me a starting point. So far I have only used the generic EQ setting, and entered the filters manually into the Winamp plug-in.

JohnPM

01-21-06, 07:22 AM

The peak search function is restricted to lower frequencies as it is looking for modal resonances. Above a few hundred Hz the resonances overlap to such an extent that it si no longer useful to try and correct individual resonances, but you can of course apply broader filters to even out the response. Future versions of the Wizard will address other parts of the range, but that is quite a way off as there is much yet to do for the low frequency range. It is worth bearing in mind that room treatments work very well in these ranges.

Darren Wadsworth

01-22-06, 05:21 PM

John,

Thank you. Very concise.

Is there anyway that a filtered sweep could be played and measured through Room EQ Wizard. I use a PC for all of my music playback. If not, would it be at all possible for you to add the capability of playing a sweep with the filters inplace, and record the results?
Currently the only way I know of to accomplish a "comparison" sweep of before and after equalization is through the use of a hardware EQ like the BFD.

Thank you
Darren

Trymor

01-22-06, 06:16 PM

Darren,

I have been using 2 PC's. I use the 'measurement PC' line OUT to the 'playback PC' line IN, and use a line-in plug-in for Winamp to hear the results through the parametric EQ, and my SPL meter/mic on the measurement PC to see the measured results . If I wanted to see the results directly, I could just connect the line OUT on the playback PC to the line IN of the measurement PC. The impulse response is off due to the delay in the PEQ, but the frequency measurement seems fine.

Another way is to use 'Virtual Audio Cable' from http://spider.nrcde.ru/music/software/eng/vac.html in place of the measurement PC. It is a 'driver' that emulates a soundcard, and allows you to have extra virtual inputs and outputs. You could then configure REW's output to Virtual Audio Cable's (VAC) input, then VAC's output to Foobars line-in (Ive seen it mentioned somewhere), through the convolver, than plug your soundcards output back into it's line-in.

You could also install a second soundcard in your PC in place of Virtual Audio Cable or a second PC.

Of course these are just workarounds, but you can see the filtered curve in REW.

JohnPM

01-23-06, 04:17 AM

Is there anyway that a filtered sweep could be played and measured through Room EQ Wizard. I use a PC for all of my music playback. If not, would it be at all possible for you to add the capability of playing a sweep with the filters inplace, and record the results?
That would be possible, albeit a little slow due to the filter response calculation before generating the sweep. I already have a feature list entry for something along these lines, I'll move it further up the list.

praz

01-23-06, 09:39 AM

I've recently put together a new computer and am unable to get the sound card set up for use with Room EQ Wizard or EFT. The motherboard is a DFI LanParty UT nF4 SLI-DR Expert with onboard Realtek ALC850 8-channel AC’97 audio. Both programs worked fine on my old computer although a bit slow. For troubleshooting I've been using EFT. While in EFT moving a volume slider one notch either direction results in either no sound or overload while doing the level check. I have tried all possible combinations over the last few days to no avail. Inputs and outputs are configured correctly on the sound card. Line-in and line-out work for music. So I give up. It's time to buy a sound card. The Sound Blaster MP3 USB sound card has been talked about quite a lot in this thread. Is this the best card to use with this software or should I be looking at something else.

---k---

01-23-06, 09:44 AM

Hey John,

Excellent program. I just started using it, and am very impressed. I was a little slow to get everything calibrated and working, but once I did, I had no problems using it. It was very easy to figure out what filters to use and get them imputed into the BFD. I'm just very impressed. Got great inital results too.

Anyway, I've only made it to page 6 of this thread so far, so forgive me if this has been discussed:

It would be great if the filters got sorted by hz.

Also, I noticed that when you load another data set, it doesn't clear out the low frequency waterfall. I exported a bunch of graphs without looking at them, and only after assembling them into one graph in photoshop did I realize they were all the same. Ooops. I should have noticed it, but...

Again, great program.

JohnPM

01-23-06, 03:58 PM

Thanks Ryan.

It would be great if the filters got sorted by hz.
You can sort the results of the peak finding by frequency or peak magnitude (just click in the column header of the peaks table), but it could be useful to be able to sort the filters that way once they have been set up. I'll add an entry to the features list for that.

Also, I noticed that when you load another data set, it doesn't clear out the low frequency waterfall.
Oops! Same for the spectral decay data, fixed that now so will work correctly in the next release.

If you have some ideas for features to add, let me know. I'm currently playing around with some energy-time curve displays so might add those soon.

Darren Wadsworth

01-23-06, 04:06 PM

John,

Thank you as always.

Trymor,

I was going through the other PC (analog in. while using REQW). But, I was unsure of how using 2 soundcards in the chain would effect the measurement (detrimental?). I am going to look for a line-in plug-in for foobar.
What are the names of the plugins you use for Winamp. Convlolver? Line-In?

Thank you
Darren

---k---

01-23-06, 04:53 PM

Thanks John.
Its really impressive how quickly you jump on bug fixes and how receptive you are to feature ideas. It is appreciated.

Trymor

01-23-06, 05:36 PM

Darren,

I use a plug-in called line_in that I found on the Winamp site under plug-ins. I also use the Shibatch Super Equalizer I found there. It is an infinite band PEQ and a graphic EQ. I haven't taken the time to do full convolution yet.

If your soundcards are good, the only frequencies really affected would be under 20Hz, and over 17KHz. Plus you can use soundcard calibration on one PC, and if you are really worried, you can do a soundcard calibration on both PC's and merge them to use on the measurement PC.

Trymor

01-23-06, 05:38 PM

John,

As long as you are taking feature suggestions, how about phase? It could help in determining the subwoofer to main crossover (and I wouldnt mind seeing phase for the entire frequency response ;) )

oliverlim

01-24-06, 08:42 PM

John,

I have a BFD and a SMS-1 as well. Due to the ability to switch between memory modes automatically as well as the crossover in the SMS-1 allowing me to more smoothly avoid any dips at the crossover section, I decided to use the SMS-1. Is I want to use your software to actually do the measurements as I want to also ensure that the EQ I apply will be at the correct point, what and how should I measure and apply it? I am hoping that the freq for BFD and SMS-1 is the same but am not sure how to convert the Q on the SMS-1 to the bandwidth on the BFD. Would appreciate any suggestion or help.

Thanks
Oliver

Bing

01-25-06, 12:59 AM

Can someone help me out with my thought process here? It's been bugging me for past 2 nites.

I want to use my mains as "small", crossed off at 80hz. I am under the impression that I should measure the sub and the mains together, right? Therefore, the FR is a combination of the main and the sub playing together. My dilemma is this: Constructing a flat FR for the sub playing by itself is easy. But once I measure both main and sub together, then it wrecks havoc. I play with xover, polarity, phase and the BFD some more so that I get a flat FR again but then I realize the sub's FR is not flat anymore. So does that mean the LFE channel is not flat?

And worse.....if I do the R main + sub and get it flat, the L main + sub is not. And certainly not center + sub. Which channel do I use? Should I just EQ the sub by itself and just ignore how the bass from the other channel combines with it? I'm almost contemplating experimenting with bookshelves as mains. Because a floorstander crossed off at 80hz still puts out more bass than I'd like. And my room likes to jack up the 40-55hz zone :(

JimP

01-25-06, 01:09 AM

Bing,

I equalize my sub with both front LRs running. I think that would more closely average the settings than most other configurations.

As to problems getting the combination of speaker and sub flat. Sounds like your speaker(s) are not set for small and you're duplicating some of the sub's frequencies. Keep in mind that depending on your speakers and your personal taste, this might be a good thing. Don't assume that a perfectly flat frequency response is always desired. I think some theaters use house curves where certain frequiencies are boosted for better effect.

JohnPM

01-25-06, 04:09 AM

As long as you are taking feature suggestions, how about phase? It could help in determining the subwoofer to main crossover (and I wouldnt mind seeing phase for the entire frequency response ;) )
It can be done, but it is a bit awkward because of the lack of an absolute time reference. Easiest is probably to generate the minimum phase that corresponds to the measured frequency response and use that, but can also make an estimate of the time delays and correct the measured phase with that (I did that a couple of months ago in a dev build and it gave reasonable-looking results). Phase displays are on the features list already but I'll take this as a vote for moving them up the list.

JohnPM

01-25-06, 08:32 AM

Bing, strictly speaking the way to address that problem is to EQ the mains at the low end - if you measure them on their own (sub turned off) you will probably find peaks in their response in the 40-55Hz range you mention despite the action of the bass management crossover. As few systems have EQ capabilities for the main channels (or a separate EQ of adequate quality) you are probably best looking at the effect of small changes in the position of the mains or your seating or both to try and reduce the peaks and focus your main EQ efforts on the sub's response. If you fancy experimenting with EQ on the mains there's nothing wrong with feeding one channel through the BFD (with the other BFD channel used for the sub) and seeing how it sounds and how well the EQ works out. A pair of BFDs would allow you to EQ the low end for sub, L, C and R, but reservations about feeding the main channels through a BFD would be understandable.

JohnPM

01-25-06, 08:47 AM

I have a BFD and a SMS-1 as well. Due to the ability to switch between memory modes automatically as well as the crossover in the SMS-1 allowing me to more smoothly avoid any dips at the crossover section, I decided to use the SMS-1. Is I want to use your software to actually do the measurements as I want to also ensure that the EQ I apply will be at the correct point, what and how should I measure and apply it? I am hoping that the freq for BFD and SMS-1 is the same but am not sure how to convert the Q on the SMS-1 to the bandwidth on the BFD. Would appreciate any suggestion or help.
Frequency is the same, I believe the SMS-1 has 1Hz frequency steps so you are probably best selecting "TMREQ" as the equaliser type on the Wizard as that also uses 1Hz steps. The Q figures on the TMREQ setting should correspond to those used by the SMS-1, if you want to see what a BFD filter setting would correspond to for SMS-1, just select DSP1124P as the equaliser, enter the filter settings, then change the equaliser to TMREQ and the settings will be updated accordingly. I can easily add an equaliser setting that corresponds to the SMS-1 to emulate the 8 EQ filters it offers, should that be of interest.

ebr

01-25-06, 09:37 AM

...but reservations about feeding the main channels through a BFD would be understandable.

Why?

Trymor

01-25-06, 10:07 AM

ebr,

I believe the biggest reason around the forum is that it 'pollutes' the systems phase response. But as far as I know, it uses standard IIR filters, so it should be no different than putting any other IIR EQ (analog or digital) in the signal chain.

On that subject, does anyone know of any (analog or digital) EQ's that use FIR filters and are inexpensive?

ebr

01-25-06, 10:09 AM

Thanks, Try.

Trymor

01-25-06, 10:19 AM

Bing,

Personally I always start with my sub (amp) crossed at 40Hz (or the lowest possible frequency), try different phase settings, and work up from there. A lot of times you will find a decent response even with the mains crossed at 80Hz. If nothing is working for you, you could try crossing the sub at 180Hz (or the highest setting) letting the receiver cross the sub over, and observe the results. Remember that your sub is being crossed over twice - once in the receiver, and once at the sub amp (if it is built in). If you can disable one of the crossovers, you should have better luck integrating everything.

A recommendation by some is to set your mains to fullrange and disable the subwoofer in the receiver (effectively sending the LFE channel to the mains). You then connect your receivers speaker outputs to your subs speaker level inputs, then your subs speaker level outputs to your main speaker inputs. This gives you only 1 crossover in the bass range, supposedly making integration easier. This also depends on your subs capabilities.

Good Luck!

JohnPM

01-25-06, 10:40 AM

but reservations about feeding the main channels through a BFD would be understandable.
Why?
Because the BFD suffers from low level mains harmonic components in its output and one or two small spikes in the noise floor at higher frequencies, plus its noise floor overall isn't as low as one might hope. The degradation in signal/noise is undetectable when feeding a subwoofer as most is outside the bandwidth of the sub, but it might be noticeable on the main channels if using sensitive speakers.

Trymor, there is nothing at all wrong with IIR filters and their phase response for countering room modes, on the contrary it is necessary to use IIR as otherwise the filter's phase response would not be opposite to that of the mode, which itself behaves like a 2nd order IIR system.

ebr

01-25-06, 10:43 AM

And thanks, John. I guess I'll stick with my Bijou for the main channels.

johnbomb

01-25-06, 11:07 AM

Hey John,

I'm having a bit of trouble setting up a series of curves for averaging. I've read the help file for that a few times, but I seem to be missing something.

JohnPM

01-25-06, 11:18 AM

I'm having a bit of trouble setting up a series of curves for averaging. I've read the help file for that a few times, but I seem to be missing something.
That doesn't give me very much to work with :), what is happening/not happening compared to what you were expecting?

Trymor

01-25-06, 11:42 AM

John,

I know the RS SPL meter has been gone over many times, but I read something that threw me a little. I found this on the ETF website:

"This unit has weighting curve adjustments (switch) that confuse a lot of users. This adjustment does not control the output, it only damps the meter movement.

Download the calibration file below and open it as a microphone calibration file when using ETF with the RS unit. Please note that this is not a true "calibration file" because it is not calibrated for your unit. It will remove the gross response errors associated with this device. Typical differences between this unit and any random unit are within +/- 2 dB below 8 KHz, typically. This unit is virtually useless for accurate high frequency measurements.

Other calibration files exist on the internet for this unit, these calibration files apply when reading the meter movement, not when taking data from its RCA output jack. These units are very constant gain below 500 Hz (flat frequency response) and do not require further low frequency calibration."

They do not however, mention whether or not there software uses C weighted compensation. The manual for the RS meter shows a ruler flat response below about 1KHz, labeled C weighting.

Does this mean that no meter 'compensation' file should ever be used for measurements below 500Hz, and software should always have C weighting compensation enabled?

P.S. Your last statement to me might explain why I wasn't getting the predicted corrections using the Shibatch PEQ plug-in for Winamp - it uses FIR filters. Any chance of adding a FIR filter function mode to REW? I believe a lot of newer EQ's coming out (including current software convolution) use FIR filters.

Trymor

01-25-06, 11:53 AM

johnbomb,

Near the top right corner of the program window, choose measurement averaging from the drop down box. Select the curves you want included in the average from the list below the drop down box you just used. Click on 'Average The Frequency Responses', and the average will be shown in the Aux channel curve.

JohnPM

01-25-06, 12:53 PM

Trymor, the "Fast/slow" switch only affects the meter movement, but the C/A-weighting switch affects the line output (at least for the analog meter, per the schematics).

I don't have any plans to implement FIR filters. Second order IIR EQ filters are common to a very wide range of products (and a requirement for modal correction), there is no such commonality in other kinds of EQ filter.

Software convolvers accept an impulse response to convolve with, this response is of course of finite length so by definition can be called Finite Impulse Response, but it can correspond to an IIR filter characteristic depending on the values of the impulse response samples (which are equivalently the values of the coefficients of an FIR filter with that length). "FIR filter" is often interpreted as meaning linear phase, but that is only the case when the coefficients meet the required symmetry constraints (for real-valued coefficients they must be symmetric or anti-symmetric about the midpoint). As an aside, linear phase FIR filters are not minimum phase, and IIR filters can be constructed that have linear phase response over a wide frequency range.

GGA

01-25-06, 01:52 PM

Hi John,

Can I use the Wizard to output a sweep 15-120Hz at around 110dB? I am interested in evaluating my subwoofer frequency response as I increase the levels.

Thanks,
George
Servodrive Contrabass and two ACI Maestros

JohnPM

01-25-06, 03:41 PM

Can I use the Wizard to output a sweep 15-120Hz at around 110dB? I am interested in evaluating my subwoofer frequency response as I increase the levels.
Yes, though your SPL meter or microphone might have significant non-linearity at such high SPLs which would affect the measurement. If you calibrate the system as described in the help but with a target SPL of 95dB instead of 75dB and you adjust your system/sub volume settings to achieve 95dB SPL at the default -20dBFS measurement level, you could raise the measurement level spinner to get levels up to around 112dB (at the max -3dBFS measurement level). You would need to change the range on your SPL meter (or mic preamp if using one) to avoid clipping, so bear in mind that the level shown would be lower than actual by the amount of that adjustment e.g. if your meter was on the 90dB range during cal and you used the 110dB range for the measurement, the figures shown on the response would be 20dB lower than actual.

The sweep starts at DC and ends at twice the selected end frequency (the additional octave ensures the results are accurate within the measurement range). Sweep duration is 3.7 seconds.

You should wear hearing protection and such levels could be very harmful to drive units, particularly the initial very low frequency section (the sweep reaches 15Hz after about 1.3s if the end frequency is set to 120Hz). To reduce the time at low frequencies, use a higher end frequency. Make measurements at much lower levels beforehand to be sure the system is set up correctly and all is working well. Examining the impulse response at time<0 would give an indication of the distortion levels.

Trymor

01-25-06, 07:03 PM

John,

So if we have the switch on the RS analog meter set to C weighting, the line output will be flat from 500Hz on down to 20Hz. Why then, are we supposed to select C weighting in REW? Wouldn't that give us a non-flat responce?

This means that the maker of ETF is totally wrong. The weighting switch affects both the meter movement and line-out. Calibration files do apply when reading the meter movement, AND when taking data from its RCA output jack.

About filters, is there an easy way to make a (or multiple) FIR flter on a PEQ emulate IIR filter characteristics?

JohnPM

01-25-06, 08:14 PM

So if we have the switch on the RS analog meter set to C weighting, the line output will be flat from 500Hz on down to 20Hz. Why then, are we supposed to select C weighting in REW? Wouldn't that give us a non-flat responce?No, when the switch is set to C weighting the line output follows the C weighting curve baceause the signal from the mic passes through the C weighting network on its way to the line output.

This means that the maker of ETF is totally wrong. The weighting switch affects both the meter movement and line-out. Calibration files do apply when reading the meter movement, AND when taking data from its RCA output jack.Yes

About filters, is there an easy way to make a (or multiple) FIR flter on a PEQ emulate IIR filter characteristics?You mean on the Shibatch? I would guess not. For a convolver you just export the filters IR from REW and convolve the audio signal with that to get the desired result.

oliverlim

01-25-06, 08:52 PM

catapult

01-25-06, 09:29 PM

This means that the maker of ETF is totally wrong. The weighting switch affects both the meter movement and line-out. Ditto with John on that. I've looked at the schematics and the C-weighting clearly applies to the RCA jack as well as the meter. That said, RS has sold many different meters with the same model name. I suppose it's possible than one of their many varieties acts as ETF describes, but I doubt it.

WarnerL

01-25-06, 09:34 PM

I've played around with the REW a bit and would like someone to interpret some findings with me having to do with the RS digital SPL meter and its line output jack.
I hooked everything up as required.

I ran the Set Measurement Level calibration choosing the subwoofer test tone when given the choice and increased the volume on my AV receiver until my RS digital SPL meter read 75 dB.

I ran the Set Input Volume calibration and the program automatically set an appropriate input level.

I ran the Calibrate program choosing Sub Cal when given the choice (used to calibrate the Wizard's SPL reading to my RS digital SPL meter reading). I didn't have to alter anything as my RS digital SPL meter read exactly 75 dB which was the default setting in this calibration program.

If I run the Set Target Level the reading in the REW reads around 75 dB which is the same reading on my RS digital SPL meter.

Now if generate a sine wave at 20 Hz with the REW my actual reading on the RS digital SPL meter happens to read 75 dB but the reading in the REW displays a level of about 6.5 to 7 dB higher, i.e. 81 dB or so. Now this is with the Compensate for C-Weighting option turned off. If I turn that option on, then the REW reading reads about 11 to 12 dB higher than my actual meter reading.

Does this at all indicate that my RS digital SPL meter line output is not outputting a c-weighted compensated signal as it supposedly does with the RS analog SPL meter?

JohnPM

01-26-06, 04:00 AM

John,

You might have missed this. Any advise? =p

Thanks
Oliver
See post #435

JohnPM

01-26-06, 04:13 AM

I've played around with the REW a bit and would like someone to interpret some findings with me having to do with the RS digital SPL meter and its line output jack.

<snip>

Does this at all indicate that my RS digital SPL meter line output is not outputting a c-weighted compensated signal as it supposedly does with the RS analog SPL meter?
Interesting. One explanation could indeed be that the meter's line output has more LF content than the internal signal the meter uses to generate its reading, so the REW SPL figure is being calibrated lower than it should be for the signal content it is being fed, hence the higher figure with your 20Hz tone. Try the same test, but calibrate the REW SPL reading using a cal tone from one of your main speakers rather than a sub cal tone and let us know the results.

oliverlim

01-26-06, 04:13 AM

See post #435

My bad. Dunno how I missed that. Sounds really good. I should have some free time to test over this long weekend. I read one another post you made about how the Q and the exact freq used for EQ is important as missing the correct freq will mean that you do not correct the ringing. That could explain why with a broad peak that I have of about 10-20db from 25hz to 45hz, I get different sounding effect with the same end graph result (not looking at the waterfall)with different EQ settings.

Great work on REQ and looking forward to your new enhancements you will be implementing.

Oliver

Trymor

01-26-06, 09:43 AM

John and/or catapult,

Why does the manual for the 33-2050 analog RS meter show a flat response graph with C weighting engaged if the units output (and meter) follows the C weighting curve? I just read further, and the manual says this: "(when using the output jack) Note that the METER response will not be flat, due to the A and C weighting networks.

That makes it sound like only the meter is affected by C weighting. But looking at the schematic in the same manual, the meter and jack are tied together.

Not only that, some people say the correction needed (if at all) for the line out is different from the meter correction. How can this be if the line out and meter are tied together?

Just trying to come to a definitive conclusion, but it is difficult seeing as the manual for it contradicts itself, along with different people.

johnbomb

01-26-06, 10:10 AM

That doesn't give me very much to work with :)

Woops, I feel like a moron. I posted that when I was away from REQW. I think I was having trouble finding the pulldown menu in the upper right part of the screen (I didn't have my glasses on at the time :D ). Well, I found it (thanks, Trymor), and averaging works like a charm. Thanks for adding that feature!

John

Darren Wadsworth

01-26-06, 05:19 PM

JohnPM

01-26-06, 06:49 PM

I am using a ECM8000 mic. Directly above the listening position (about about 4 feet is a CRT project on ceiling). Should I point the mic straight up or, at say, a 45deg angle toward the front center of the room? Between the 2 mains. Would the projector interfere with the readings?
Point the mic straight up for low freq measurements, or at the speakers for full range measurements. Projector won't be a problem as long as it's off :) otherwise the fan(s) add to the background noise.

Darren Wadsworth

01-26-06, 10:32 PM

Thanks John.

I not only turned off the projector, but turned off the central heating and unplugged the fridge (its just outside a doorway at the back of the room) :)
When I did that it was so quite in the room that I heard a never before heard hum from inside the power amp in the sub. :)

Darren

AndrewS99

01-28-06, 11:46 AM

John:
Quick question. I got an Outlaw 990 and I'm trying to EQ my subs with Room EQ Wizard. Had no problem as far as I could tell on my B&K Ref 31 but when I run a sweep from 20Hz to 200Hz using the digital out on a Creative SB MP3+ (optical) the sweep signal goes to the subs correctly but also goes to the center channel - which I don't think is right. Does this sound like an Outlaw 990 issue or did I set something incorrectly in Room EQ Wizard?

Room EQ Wizard is AWESOME by the way. I've done a ton of playing around and am in the process of anal retentively documenting the setup with a laptop, Creative SB MP3+ sound card and a Behringer ECM8000 mic. I'll post it when I'm done...

JimP

01-28-06, 12:21 PM

Andrew,

I'd be looking at your preamp/receiver to see if its setup to take a stereo signal and distribute it across your front 3.

JohnPM

01-28-06, 02:06 PM

John:
Quick question. I got an Outlaw 990 and I'm trying to EQ my subs with Room EQ Wizard. Had no problem as far as I could tell on my B&K Ref 31 but when I run a sweep from 20Hz to 200Hz using the digital out on a Creative SB MP3+ (optical) the sweep signal goes to the subs correctly but also goes to the center channel - which I don't think is right. Does this sound like an Outlaw 990 issue or did I set something incorrectly in Room EQ Wizard?
If you have connected both L and R to the processor then the behaviour will depend on the mode you have set for the input you are using, if you have it set to PLIIx or 5-CH STEREO, for example, the signal will be treated as mono (as L and R are the same) and routed almost entirely to the centre channel, with LF content redirected to the sub according to the bass management settings. If you select "Stereo" you should get L, R and Sub only.

mlbrand

01-28-06, 02:14 PM

AndrewS99,

I think you may have a bass management issue ("bug") with your Outlaw 990. Read this thread AVS-More issues with990, audio related (http://www.avsforum.com/avs-vb/showthread.php?t=636738)

Trymor

01-28-06, 05:58 PM

John,

Thanks again for adding the Ctrl-3 shortcut, it has made trying different speakers and positions TONS easier.

Is there a reason that REW doesn't assign a filter at 155Hz in this response graph? When I try a different set of speakers in almost the same spot, the peak shifts down to 125Hz and is taller, but REW doesnt assign a filter there either. Both peaks are around 10db, I have the 2db removal option off, and I am using the generic EQ option.

AndrewS99

01-28-06, 06:51 PM

mlbrand:

The bass management "bug" comes from analog connections - I was using a digital connection from the sound card to the pre. There is definitely no stereo "front 3" mode - but John may be on to something. I may have had it set in PLII mode and not stereo . I'll have to go back and check and see if it does this in stereo.

John:

I got my dual subs sounding GREAT and looking mostly "hump free" on a preloaded house curve - thanks again for this great software. It's invaluable for my system. :)

JohnPM

01-28-06, 07:13 PM

Is there a reason that REW doesn't assign a filter at 155Hz in this response graph? When I try a different set of speakers in almost the same spot, the peak shifts down to 125Hz and is taller, but REW doesnt assign a filter there either.Depends on the details of the shape of the response, the current filter assignment method is fairly picky. If you send me the mdat file including the measurements I can use it for testing future versions of the peak/mode detection algorithms - in the meantime, simply assign manual filters.

Trymor

01-28-06, 08:36 PM

John,

Its no big deal, I was just curious. I already assigned a bunch of filters manually, and overwrote the mdat file. If you requested it to help me, thanks, but I'm set. If you requested the file because it would be really helpful to you, let me know and I will re-create it.

I decided to give up using the FIR filter PEQ and am giving convolution a shot, since I changed speakers. I am using RealReverb for winamp (since I couldn't get the Convoler plug-in to read the wav files), and I am wondering why REW's wav files are so long. I am using an IR size limit of about .2 seconds in RealReverb because it reduces latency greatly. The sound doesn't seem to be different than when I let it use REW's full IR wav file. Am I missing anything by limiting the file?

Tukkis

01-29-06, 04:44 AM

By importing a calibration mic correction file, should I be trying to match the freq. response to the mic calibration curve or the normal target response?

Trymor

01-29-06, 05:37 AM

Normal target response.

Tukkis

01-29-06, 05:43 AM

Ok so the mic calibration curve is showing an inverse of the corrections it's applying?

JohnPM

01-29-06, 09:15 AM

Ok so the mic calibration curve is showing an inverse of the corrections it's applying?
The mic cal curve is subtracted from subsequent measurements, so it should be the actual response of the mic.

JohnPM

01-29-06, 09:19 AM

Its no big deal, I was just curious. I already assigned a bunch of filters manually, and overwrote the mdat file. If you requested it to help me, thanks, but I'm set. If you requested the file because it would be really helpful to you, let me know and I will re-create it.

I decided to give up using the FIR filter PEQ and am giving convolution a shot, since I changed speakers. I am using RealReverb for winamp (since I couldn't get the Convoler plug-in to read the wav files), and I am wondering why REW's wav files are so long. I am using an IR size limit of about .2 seconds in RealReverb because it reduces latency greatly. The sound doesn't seem to be different than when I let it use REW's full IR wav file. Am I missing anything by limiting the file?
The mdat file was for my benefit, if you get a chance to make a new one send it on to me.

The filter IR's are 128k samples, whether truncating them makes a difference depends on how far the response has decayed. The next release (prob next week) allows you to import as well as export WAV impulse responses, so can load the response back in to look at it and see whether truncating to 0.2s leaves you with the freq response you want by applying a rectangular window of that width to the response and loloking at the resulting freq response.

3no

01-29-06, 10:06 AM

I'm just getting started with REW. First steps go OK. Soundcard calibration looks OK in that I get a nice curve which looks similar to the help file (except it is based at 75db instead of 0db as in the help file). However, when I run the check calibration (on either channel), the sweep *exactly* overlays the dotted calibration curve, as if the soundcard calibration compensation is not being applied. I believe that I followed the instructions, saving the cal file, turning off C weighting on the meter menu.

I'm sure I am doing something wrong, but I can't figure it out from the help files or from searching this thread. I have attached screen shots of the cal results and the cal check results, the cal file and the soundcard debug file, if any of those would help.

Mark

Tukkis

01-29-06, 10:08 AM

The mic cal curve is subtracted from subsequent measurements, so it should be the actual response of the mic.

Ok but it autocorrects the freq response according to the mic cal file right?

So you dont need to display the mic freq response for anything? It just shows what the mics doing.

JohnPM

01-29-06, 12:54 PM

Ok but it autocorrects the freq response according to the mic cal file right?

So you dont need to display the mic freq response for anything? It just shows what the mics doing.
Correct

JohnPM

01-29-06, 12:57 PM

I'm just getting started with REW. First steps go OK. Soundcard calibration looks OK in that I get a nice curve which looks similar to the help file (except it is based at 75db instead of 0db as in the help file). However, when I run the check calibration (on either channel), the sweep *exactly* overlays the dotted calibration curve, as if the soundcard calibration compensation is not being applied. I believe that I followed the instructions, saving the cal file, turning off C weighting on the meter menu.
You need to select a channel other than the soundcard channel for the check measurement, the soundcard cal is not applied when measuring the soundcard channel (i.e. when the soundcard tab is selected in the filters panel).

3no

01-29-06, 02:23 PM

Darren Wadsworth

01-29-06, 08:29 PM

Please help me with a question regarding Speaker type for setting filters. I am taking seperate readings from right and left mains with sub turned on.
What "speaker type" setting should I use for the "slope" to determine the peaks for filtering?

"The Speaker Type for each channel can be set to "Bass Limited" (often referred to as "Small"), "Full Range" (often referred to as "Large"), "Subwoofer" (strictly speaking only applicable for the Sub channel) and "None""

Thank you
Darren

JohnPM

01-30-06, 12:08 AM

Please help me with a question regarding Speaker type for setting filters. I am taking seperate readings from right and left mains with sub turned on. What "speaker type" setting should I use for the "slope" to determine the peaks for filtering?
When taking readings with the sub on the aim is for an overall flat response, so set the speaker type to "Full Range".

Tukkis

01-30-06, 01:02 AM

JohnPM

01-30-06, 04:09 AM

When trying to calculate filters for Eqing, should you take a measurement of just the sub or the sub with the mains?

I'm thinking around the crossover region 60-100hz there would be an overlap that could cause a peak and would add a few db to the plot that wouldn't be there with just the sub? Or is this something that should be fixed with phase?
See posts #431 (http://www.avsforum.com/avs-vb/showthread.php?p=6991994&&#post6991994) & #434 (http://www.avsforum.com/avs-vb/showthread.php?p=6992761&&#post6992761)

Tukkis

01-30-06, 06:55 AM

Thanks John

Not sure if it's been mentioned but is there a way to add notes to each channel?

Eg. Putting sub placement details for each channel so you know what positioning you used (1ft from front wall, left front corner)

JimP

01-30-06, 07:01 AM

Tukkis,
Name the jpeg.

Tukkis

01-30-06, 07:02 AM

Tukkis,
Name the jpeg.

That could work.

I just wasn't sure if something had been implemented for taking notes and I couldn't see it.

JohnPM

01-30-06, 07:27 AM

There's no per-channel notes facility but it's a good one to put on the features list, will need some thought on how to integrate it nicely in the GUI.

You can enter notes when you save a set of measured data and those are visible on the preview in the file chooser and in the View -> Measured Data Summary dialog after loading a set of data. The name for the channel also allows a few chars to be added as shorthand, the names on the tabs can be longer than those shown on the graphs.

Trymor

01-30-06, 08:18 AM

John,

Here is the data set and filter settings to look at. The tab renamed to System is a pair of Definitive Technology BP2X bi-polar speakers with the ports plugged. They are used as mains on an open shelf at ear level, with stereo 10" plugged port subs on the floor.

The tab renamed to Logitech is a Z640 system angled up from the desktop, with the integrated sub moved to the floor in front of one of the same 10" subs.

This is the first time REW assigned a filter for the peak at 125Hz on the Logitechs, but it still didn't assign one for the 155Hz peak for the BP2X's.

You seem like a patient guy answering all these user questions, and your GUI design is pretty logical. Do you have any other software you have programed?

Turns out I will have to E-mail the files to you. The zipped file is to big for the forum to accept.

JohnPM

01-30-06, 07:37 PM

Trymor, no other s/w from me in the public domain bar a little Java app I wrote to show Rio Karma users how the player's filter settings affect the frequency response. 25 years of software writing behind me though (but I still haven't figured out how to leave out the bugs ;))

Tukkis, I've had a play with ways of entering/displaying notes tonight and think I've come up with a workable implementation, so I'll include it in this week's release.

dknightd

01-30-06, 07:52 PM

HMenke

01-30-06, 08:51 PM

You need to select a channel other than the soundcard channel for the check measurement, the soundcard cal is not applied when measuring the soundcard channel (i.e. when the soundcard tab is selected in the filters panel).

John, I am getting started with your fine software tonight and ran into this same issue. After completing all previous steps, with the Right soundcard output looped back to the Right soundcard input, I executed the "measure soundcard response" function and generated a nice response curve similar to the one shown in your help file.

Without changing any physical connections (no necessary changes were mentioned) I clicked to the Left tab, turned off C weighting, and clicked Automatic Measurement. I don't exactly understand what is happening here. What is being measured? Nothing is connected to the soundcard's left channel, In or Out. No sound is emanating from the Left speaker because nothing is connected to the A/V receiver at this point. The only red line I get is a little low-amplitude blip above 24 kHz (see attached image). What am I doing wrong?

Thanks,

Henry

Edit: May have solved my own problem. I realized that the red blip may be part of a curve. Then I noticed the slider on the left side and brought the red curve up into view. Does this look OK? What is the other curve rolling off below 300 Hz? I didn't see this curve in the help file. Thanks, Henry

Exocer

01-31-06, 12:03 AM

Hey,
Im just finishing up running some sweeps of my room and have a few graphs (which i will post tomorrow). I actually have no idea how to apply the correction settings to my analog RS spl meter, so all my measurements are probably a farcry from accurate. Somehow, testing my bookshelfs alone, my graphs are showing a drop of output at 50hz (which is spec for my speakers) this drop goes until about 30hz,then a major boost in output from the 20hz-30hz range appears on the graph... this is without the subwoofer being tested.

Connection method: Sound card used is the Chaintech AV710
SPL meter is connected to the line-in on my sound card, with a mini-rca y adapter. Only the right channel connected to spl meter.
My AVR is connected to a line-out on my sound card. Right side connected to AVR.
I selected my sound card in the both input and output devices, and calbrated them according the instructions. Are my connections okay?

The listening space is very small, 13'x10' with 8 ' ceilings. I want to listen at around 75 db. 85db for movies, 75 db for music. Reading on this forum i've found that having a small listening space like mine can cause increased nulls and/or peaks, which is why I want to plot my room FR. If for any reason my graph is decent, I wont drop the extra cash for a BFD that I wouldn't even need.

JohnPM

01-31-06, 04:37 AM

Hi,
Has anybody had any luck getting this to run under Mac OS x?

When I try to calibrate my soundcard I get

"Impulse peak is not where it should be,
the measurement may have been corrupted

Check the Impulse Response and the Captured
Data plots"
A few hardy souls have managed it. You need to set levels manually on the Mac using the Mac's level tweaking tools as there is no programmatic access for the Wizard. When you look at the Captured Data plot is it seeing any signal from the loopback? Did setting the measurement level and calibrating SPL go OK?

JohnPM

01-31-06, 04:41 AM

Edit: May have solved my own problem. I realized that the red blip may be part of a curve. Then I noticed the slider on the left side and brought the red curve up into view. Does this look OK? What is the other curve rolling off below 300 Hz? I didn't see this curve in the help file. Thanks, Henry
Looks OK. Did you calibrate at 72dB rather than 75? The blue curve is the target response for a bass limited speaker with an 80Hz cutoff (hence -3dB at 80Hz).

sensibull

01-31-06, 07:11 AM

I actually have no idea how to apply the correction settings to my analog RS spl meter, so all my measurements are probably a farcry from accurate. Somehow, testing my bookshelfs alone, my graphs are showing a drop of output at 50hz (which is spec for my speakers) this drop goes until about 30hz,then a major boost in output from the 20hz-30hz range appears on the graph... this is without the subwoofer being tested.

REW applies its own correction curve for the RS spl meter. The boost you see in the 20-30Hz could be room gain, likely in a space that small.

dknightd

01-31-06, 07:32 AM

A few hardy souls have managed it. You need to set levels manually on the Mac using the Mac's level tweaking tools as there is no programmatic access for the Wizard. When you look at the Captured Data plot is it seeing any signal from the loopback? Did setting the measurement level and calibrating SPL go OK?

Calibrating the measurement level and SPL went fine. When I try to a calibration
of the soundcard a graph is generated, so it appears to be seeing data in the loopback.
I noticed that before the sweep begins that there is a "blip" in the audio.
Or maybe POP is a better word for it. Perhaps the analysis is trying to key
off this POP instead of the calibration sweep? I captured the soundcard input
in another audio program that lets me look at the waveform, and
apart from this POP it looks like a swept sine wave as expected.
When I tried to do an automatic
measurement, without calibrating the card, the same POP is heard, the same
error messages is given, but the program crashes when I hit OK.
I've tried both an m-audio audiophile firewire, and an older Edirol USB
card with the same result. I don't usually gets pops on my audio.

I suspect the problem might be with the Java sound output device.
I notice, on a mac, that the program uses core audio for input but
java sound engine for output. Is there anyway to have it use core audio
for the output as well?
I'm going to start searching for a patch to je5 that might effect sound output,
unless you have a better idea.

Thanks
David

oliverlim

01-31-06, 09:15 AM

Hi John,

I just spend 3 hours playing with REW! =p A little tough in the beginning but got the hang of it I hope. But ran into some issues.

1. The sound calibration, SPL calibration went ok. The Input level did not work for me. It says that my sound card did not seem to allow input level adjustment. I am using a Mobile Pre USB sound adapter. The RMS level seems to be stable at about -12db but I understand it should be at -18db for the best result. But since it seems to measure and work ok I left it alone.

2. The sweep results for REW, TrueRTA(1/24) & SMS-1 all seems rather differnt. I am using the velodyne mic and use the EM8000 for the TrueRTA and REW. Even the 200hz to 20Khz results seems rather differet between REW and TrueRTA. The REW one seems more all over the place somehting like +-12db at 1/24 from 200hz up but the TrueRTA one at 1/24 is more like +-6db. I am not sure if it is some noise issue. The Low freq results is more similar but even then, there are dips at 57hz that is not picked up by both SMS-1 and the TrueRTA(1/24).

3. Also when ever I exit REW, the sound that comes out from my speakers is loud and nuts. I am worried I blew my speakers! =p Could that be the reason I am getting different results from REW?

4. Using the TEMQ EQ, the EQ when applying to SMS-1 does not get me even anywhere near what the predicted results. Again, the 57hz dip is driving me crazy. Moving a EQ away in 1hz from the 57hz e.g. 40hz to 39hz will sometimes make the dip at 57hz drop by 5db or more. But mving it to 45hz restores the dip by 5db. Not sure how this could happen.

I guess I must be doing something wrong. Any help would be appreciated.

Oliver

HMenke

01-31-06, 09:51 AM

Looks OK. Did you calibrate at 72dB rather than 75? The blue curve is the target response for a bass limited speaker with an 80Hz cutoff (hence -3dB at 80Hz).

I am not sure. I set the meter to 80dB and then set the amplifier volume so that the needle was reading -5dB, or 75dB. I did notice that if I switch to 70dB, the needle only goes to about +3dB, or 73dB. If I increase volume so it goes to +5dB, then when I switch back to 80dB it reads -3dB, or 77dB.

I also noticed that the SPL shown in the software window was different than the meter reading. I can't remember exactly but it seemed like it was higher, around 81dB.

On my way to work this morning I realized the genius of your calibration checking method. Since we have the output looped to input, and then tell the software that it is receiving a signal from the "LEFT" speaker, the software is fooled into thinking that the speaker's response is very flat.

Is the AUX panel intended to be used for checking the entire system response, i.e. all speakers plus subwoofer, rather than just individual channels?

Henry

JohnPM

01-31-06, 10:09 AM

I suspect the problem might be with the Java sound output device. I notice, on a mac, that the program uses core audio for input but
java sound engine for output. Is there anyway to have it use core audio
for the output as well? I'm going to start searching for a patch to je5 that might effect sound output, unless you have a better idea.
Try leaving the output (and even the input) device selectors blank (i.e. saying "choose device", then the OS should pass the app the default interfaces. Might work better. What signal level gets used during the soundcard cal test? Would be interesting to see the screen capture of the Captured Data trace. As all was OK with the measurement level and SPL cal can just skip the soundcard cal and try making some measurements.

JohnPM

01-31-06, 10:23 AM

1. The sound calibration, SPL calibration went ok. The Input level did not work for me. It says that my sound card did not seem to allow input level adjustment. I am using a Mobile Pre USB sound adapter. The RMS level seems to be stable at about -12db but I understand it should be at -18db for the best result. But since it seems to measure and work ok I left it alone.
Should be fine, though you might get clipping if there is a very strong resonance. If that happens just lower the measurement level manually using the small control on the signal generator panel.

2. The sweep results for REW, TrueRTA(1/24) & SMS-1 all seems rather differnt. I am using the velodyne mic and use the EM8000 for the TrueRTA and REW. Even the 200hz to 20Khz results seems rather differet between REW and TrueRTA. The REW one seems more all over the place somehting like +-12db at 1/24 from 200hz up but the TrueRTA one at 1/24 is more like +-6db. I am not sure if it is some noise issue. The Low freq results is more similar but even then, there are dips at 57hz that is not picked up by both SMS-1 and the TrueRTA(1/24).
Pretty much what I would expect. An RTA on 1/24 setting averages the results in "bins" that are 1/24th of an octave wide, for a similar effect on REW try setting Trace Smoothing to 1/24, then results should be closer to TrueRTA. REW is giving you a higher resolution view of the response. Don't know what measurement method SMS-1 uses so can't comment on that. Overall sounds like it is working OK.

3. Also when ever I exit REW, the sound that comes out from my speakers is loud and nuts. I am worried I blew my speakers! =p Could that be the reason I am getting different results from REW?
That's a strange one, do you mean that closing the app generates a noise, or that your volume levels are very high afterwards? When shutting down, REW restores (or tries to restore) the volume settings it found when it started up.

4. Using the TEMQ EQ, the EQ when applying to SMS-1 does not get me even anywhere near what the predicted results. Again, the 57hz dip is driving me crazy. Moving a EQ away in 1hz from the 57hz e.g. 40hz to 39hz will sometimes make the dip at 57hz drop by 5db or more. But mving it to 45hz restores the dip by 5db. Not sure how this could happen.
I came across some SMS-1 response plots in a forum post, the Q figure is interpreted differently to that in the TMREQ setting. I've added an SMS-1 setting to this week's release, so try that. The behaviour you describe seems odd though, if you save measurement data as an mdat file, zip that and email it to me I'll see if it looks like a valid measurement.