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Room EQ Wizard (free measurement and parametric EQ setup software) - Page 2

post #31 of 856
Quote:


Originally posted by sensibull
If I am reading the requirements correctly, my soundcard would need right and left (i.e. two) lines in and out to use this software? Neither my onboard sound, nor my soundcard (Chaintech AV-710) have two lines in or out, but might I be able to use the single lines in and out from both? Or could I split a single line in and a single line out?

What state is your 710 in , Mine is flashed with the newer firmware, and does not do analog, only didital.

Thanks
post #32 of 856
Thanks W4ZOO, but I never flashed mine with the Prodigy firmware, though up 'til now I have only been using the optical out (I get bit-perfect with 3.10a drivers). I was just being an idiot in reading the Room EQ instructions, and not realizing from the get go that I could split the lines in and out into right and left channels.
post #33 of 856
Extremely neat and self eplanatory program.
I must test this on my Denon AVR-3805 that also has a built in self parametrizing / equalising system using the Denon meassure microphone as in pic below


Then I will make some test from the PC and see if the result adds up or if it is way off - if it is off i can allways tune the AVr-3805 with its 8 band equalizer to get it closer to good. A Pity one aint got all those functionalities that are built into this program on the AVR-3805 though

I will start of using the Denon microphone as the base (since it is desgined for the Denon AVR-3805) - will that work fine even if its not a SPL meter it is a meassuring microphone - same thing/will work or not?

Thanks a bunch for your work and for sharing it with us

Regards
Boogieman
post #34 of 856
Thread Starter 
Quote:


Originally posted by DavidDahl
Hi all,

I've been waiting a long time for something like this.

I have also run into two problems while using it. I am convinced it's not a problem with the program but rather a problem on my end. First, the computer cannot seem to locate the loopback on the left channel. I have triple-checked the connections and am so far at a loss. Second, Although I can generate test tones that are picked up at the SPL meter I cannot seem to calibrate the program. It says that the input level is set to maximum (1.0) and that I should reduce the settings on the SPL meter. My concern is that the SPL meter is already set pretty low and I am unable to hear anything from the right speaker indicating that there is calibrating even going on.

Any ideas?

-Dave

Hi Dave,

What are the Input Device and Input set to? What soundcard are you using and how are the connectors labelled? From the description likely problem is the input you are connected to is not the one that has been selected, or there may be some other input mixer control that is preventing the wizard picking up the signal. Have you already used this setup with other software such as ETF?

Another possibility is that there is a problem on the signal generation side, again selected vs connected output, or some mixer control preventing the signal getting out. What are the outout device and output set to?

A way to check whether the external connections are the problem can be to select "stereo mix" as the input (assuming that appears in the list), that is a local loopback on the soundcard, with that selected you should be able to see the effect of generating test tones on the various readings in the spl panel if the output side is set up correctly.

Could post some screen captures of the windows mixer settings for record and replay after you have started the wizard up, get to the mixer via the Sounds and Audio Devices item in the control panel and select "Advanced" in the "device volume" box, make one screen cap with properties set to recording and another with properties set to playback and either post them here or email them to me.

Regards,
post #35 of 856
Thread Starter 
Hi Boogieman,

I'm not familiar with the Denon mic so not sure if it has mic level or line level output. If mic level it will be a problem, really need line level signals.

Regards,
post #36 of 856
John,

I finally had time to sit and play with your software. Congratulations on a very easy to use interface. I'm still scouring the help to make sure I'm doing everything correctly, but so far I'm really enjoying it. With your hopes to add midi-connection capability to the PEQ, this is just going to get better.

Thanks!
post #37 of 856
Reading this thread it and the instructions on soundcard connection, it sounds like the connections should be the same as ETF. I can get ETF to work, but wasn't able to get the RoomEQ Wizard to work. I do get a feedback loop when trying to calibrate. If I have it set to track the SPL, if I speak into the test microphone, it is output through my system. I am using a Creative Soundblaster USB soundcard, Behringer ECM8000 mic and 802 preamp/phantom power. Room EQ settings are input = LINE-IN; output = SPEAKER; windows mixer only has LINE enabled for record and PLAYBACK and WAVE enabled for playback. These settingswork for ETF but for the Wizard, I'm not getting a tone out but if I set the prepro level high enough I get a feedback loop - the mic input is going straight to the output...???
Anyone have any suggestions...???

Thanks, Wyatt
post #38 of 856
Thread Starter 
Hi Wyatt,

Only wave should be unmuted for playback, if you unmute Line as well you will get feedback as described. The Wizard mutes all playback sources except for wave when it starts up, did it not do this on your setup?

Regards,
post #39 of 856
OK, fundamental question....when doing the initial calibration is the RoomEQ generating the test tone or are you using the internal tone on your processor??? If RoomEQ should be generating the tone, what type should it be - white noise???

I get output if I turn on the signal generator section....

Thanks,
Wyatt
post #40 of 856
Thread Starter 
When calibrating the SPL meter, use the internal tone on your processor and make sure the cal level that has been set in the Wizard's SPL meter panel is the same as the reading on your handheld SPL meter. That sets the reference level for input signals, which is then used when setting the levels for measurement.

Regards,
post #41 of 856
John,

Which of the many downloads available here

http://java.sun.com/j2se/1.5.0/download.jsp

should I choose?

Do I correctly presume that I don't need to know Java and it's use is transparent to the user?

Thanks
post #42 of 856
Thread Starter 
Correct, no need to know anything about Java. You need to download "JRE 5.0 Update 2".

Regards,
post #43 of 856
noah katz
It's the "run time" version - JRE which just lets the java programs run. No need to "know" anything at all. Just updating when there are major security updates available is probably not a bad idea.
post #44 of 856
Thanks!
post #45 of 856
Getting my feet wet here...

The input device selection window on the lower left shows Avance AV97 Audio, which is what my laptop has.

Does this mean it's suitable (full duplex, etc) for Room EQ Wizard or is it just reporting its presence?

I haven't been able to get my laptop to tell me its sound card capabilities.

Thanks
post #46 of 856
Thread Starter 
The wizard lists any soundcards it finds that report they support the required formats. Doesn't necessarily mean it is suitable but most are nowadays.
post #47 of 856
Thanks, John, I'll give it a try and see.
post #48 of 856
Forgive me for asking another idiot newbie question, but in your help file you specify to use the AV processor's speaker trim to adjust L, R, or C speaker output until it matches target Cal Level dB, but to adjust the volume during the Measurement Level setting.

While I understand the difference between trim and volume, I'm a little confused about how to hit 75dB using the trim adjustments alone (which on my HK 635, only go from -10db to +10db). Is the assumption that I have previously calibrated my system and already know what volume level should hit 75dB on the test tone, and that I set the volume to that before I begin the SPL calibration, using the trim settings only to fine tune it? Or does it matter whether I use volume alone, or volume + trim during the SPL calibration phase?

While I'm already embarassing myself, I have two further clarification questions:
1. The point of the two calibrations is to match the RadioShack SPL with the Room EQ SPL (as dictated by recording level), and then to determine at what volume your soundcard creates a 75db signal, correct?
2. Does it skew the measurements if you use a lower target SPL? I have twin babies and never run my system very loud. I ran a preliminary test last night and even 80dB left my ears ringing a bit...
post #49 of 856
Thread Starter 
Quote:


Originally posted by sensibull
Forgive me for asking another idiot newbie question, but in your help file you specify to use the AV processor's speaker trim to adjust L, R, or C speaker output until it matches target Cal Level dB, but to adjust the volume during the Measurement Level setting.

While I understand the difference between trim and volume, I'm a little confused about how to hit 75dB using the trim adjustments alone (which on my HK 635, only go from -10db to +10db). Is the assumption that I have previously calibrated my system and already know what volume level should hit 75dB on the test tone, and that I set the volume to that before I begin the SPL calibration, using the trim settings only to fine tune it? Or does it matter whether I use volume alone, or volume + trim during the SPL calibration phase?

While I'm already embarassing myself, I have two further clarification questions:
1. The point of the two calibrations is to match the RadioShack SPL with the Room EQ SPL (as dictated by recording level), and then to determine at what volume your soundcard creates a 75db signal, correct?
2. Does it skew the measurements if you use a lower target SPL? I have twin babies and never run my system very loud. I ran a preliminary test last night and even 80dB left my ears ringing a bit...

Hi,

I'll tweak the help text for those sections to make it a little more clear, it is not the most intuitive part of the process.

The input calibration is used to give the Wizard an absolute SPL reference. A signal needs to be generated at a known level (as read off your SPL meter) and the Wizard needs to be told what that level is (via the Cal Level dB control). It will then adjust the input level control so that the signal captured by the soundcard has enough headroom (aims for 18dB) and adjust its own SPL meter reading to show the same figure as your SPL meter.

The process is based on the normal calibration routine for your AV processor, as carried out when you set up your system, using the internal calibration tones on your processor. The level of the calibration tone is usually not affected by your volume control, only by the speaker trims (the processor typically applies an internal volume setting that is appropriate for the calibration process). If the volume control does affect the calibration tone level, set it to whatever level is recommended for calibrating your processor. You will only need to alter the speaker trim if the level on your SPL meter is not at your calibration level - alternatively, just change the Cal Level dB setting on the wizard so that it is the same as the reading on your SPL meter.

Usually the cal level is 75dB, you can use a lower figure as long as you can trim your cal tone output to hit that lower level during calibration.

Setting the measurement level, which does involve adjusting your processor's volume control, is to take account of the output level of the soundcard and the input sensitivity of your processor and set a level for taking measurements that is high enough for accurate results but has enough headroom to allow for the effects of resonances. The target it aims for is to achieve the same level as used for the input calibration. First it sets its own output sig gen level by measuring the level on the local loopback connection, then it generates a speaker cal signal and waits for you to adjust the AV processor's volume to give the desired level measured off the SPL meter.

Regarding keeping the noise down, if you select the "Limit SPL when measuring" box below the Set Target Level button it will automatically reduce the signal levels when resonances are encountered to keep the overall level close to the target level for the speaker - it cuts the excess above the target to 1/4 of what it would otherwise be, e.g. a 12dB resonance would only give a level 3dB above the target. The graph will show the figures that would have resulted without the limiting, i.e. it would still show a 12dB peak. This does extend the measurement time though, as the limiter needs additional settling time. You can even set an artificially low Target Level during the measurement (e.g. 65dB) to reduce levels even further, as long as you remember to set the correct target level before searching for peaks or adjusting filters. However I have just found a bug in the measurement figures that are recorded when the limiter is on so wait for tonight's release if you want to use that feature.

Regards,
post #50 of 856
Thread Starter 
V3.19 has now been uploaded to the website. The main addition this time is Midi comms
for setting up BFD Pro filters, though you will also need a Midi interface. Details of
the Midi comms are in a new help file here:
http://homepage.ntlworld.com/john.mu.../bfdcomms.html

There have been various other updates in the help text to add references to BFD Pro where
appropriate.

There is also a link to download a zip file of the jars used by the wizard so the app
can be tried on Linux or Mac OS X platforms - I haven't tested on other platforms, however.

Summary of V3.19 changes:
  • BFD Pro can now be set up over a Midi connection
  • Added a "Comms" menu, put COM port and Midi input and output port selection in there
  • Tweaks to helptext for unit.html, equaliser.html, soundcard.html, welcome.html (link to download site for J2SE V5.0 for Max OS X), avpcomms.html
  • Tweaks to the English help text for makingmeasurements.html, inputcal.html, filteradjustment.html
  • New helptext files added: comms.html, bfdcomms.html
  • "Generate Soundcard Debug File" added to the Soundcard menu to generate a text file with debug info following reports of problems with some multi-channel soundcards (RME 9632)
  • When the equaliser is BFD Pro the Aux tab is renamed to Sub2
  • Fixed a bug that caused wrong measurement values to be recorded when the SPL limiter was active

Regards
post #51 of 856
Are the widely published correction values for the Radio Shack meter and Room EQ's C-Weighting Compensation adjustments essentially the same thing, or totally unrelated?

In other words, if I am importing measurements manually (with a comma delimited txt file) that were made with a C-weighted Radio Shack meter and that have had the correction values already applied, do I still need to apply the C Weighting Compensation, or would that be redundant?

Also, can Room EQ suggest filters for correcting dips or valleys as well as peaks? I know boosting signals is generally frowned upon, but it may be called for in my situation...

Thanks again for you continued assistance...
post #52 of 856
Thread Starter 
Quote:


Originally posted by sensibull
Are the widely published correction values for the Radio Shack meter and Room EQ's C-Weighting Compensation adjustments essentially the same thing, or totally unrelated?

In other words, if I am importing measurements manually (with a comma delimited txt file) that were made with a C-weighted Radio Shack meter and that have had the correction values already applied, do I still need to apply the C Weighting Compensation, or would that be redundant?

Also, can Room EQ suggest filters for correcting dips or valleys as well as peaks? I know boosting signals is generally frowned upon, but it may be called for in my situation...

Thanks again for you continued assistance...

The various RS meter correction values tend to be a mixture of the required C-weighting compensation and corrections for individual meters that may or may not be valid for others.

If you are importing data to which you have already applied corrections in the range below 200Hz, safest is to say "no" when the Wizard asks if you want C-weighting compensation applied. Alternatively, if you know the corrections that were previously applied you could manually reverse them and then import with C-weighting compensation, but probably not worth bothering.

The automatic EQ routines only search for peaks in the response, and only apply filters to correct those peaks. You can manually apply filters to boost portions of the response, but obviously don't waste any time trying to fill notches/nulls.

Regards,
post #53 of 856
Thanks John!

I tried out your new version with my BFD, Audigy 2 ZS and RS
analog meter. Had to buy a couple of MIDI cables just to try out the
sending of the filters to the BFD. It all worked! I admit to adding one
more notch and a couple of mild peaking filters as manuals after the
auto optimize initial filter set.

I especially noticed that the left channel feedback calibration really
made a difference in the 10 to 20 Hz measurements with the Creative
Labs Audigy 2 ZS. Just like the similar significant results I got when I
ran TrueRTA's soundcard calibration.

BTW I checked my equalized results with TrueRTA and they seemed to
jive. I probably could have skipped paying $100 for TrueRTA if I had
waited for your Room Eq Wizard software.
post #54 of 856
Thread Starter 
Glad to hear all went well. Manual tweaking of filters is very much encouraged, the auto setting is a convenience feature to deal with the main problems, further optimisation is then in the hands of the user.
post #55 of 856
Thread Starter 
The Wizard has now reached V3.20, changes this time:
  • Distortion level displays have been added to the SPL Meter panel, showing the levels of the 2nd and 3rd harmonics of the test frequency. There are separate displays for the local loopback from the soundcard and for the external input - bear in mind that high noise levels will show up as correspondingly high 2nd/3rd harmonic levels. The SPL Meter panel has been rearranged to add the new displays, also rearranged the signal generator panel a little to keep its appearance consistent
  • After rearranging the SPL Meter panel there was some empty space on the RHS, so I decided to code up some level meters to go there. Don't tell you anything you didn't already know from the numeric peak and rms displays, but does add some colour
  • I've tweaked the SPL limiter loop to reduce settling time
  • Bug fix: SPL calibration value was not being saved if the soundcard had no input volume control (e.g. some USB cards)
  • Added help index files in English and Dutch help directories and links to the index and the home page at the bottom of each help file - mainly to provide links back to somewhere central for the web version of the help pages
  • Added notes on optimiser Q/BW limits in filter adjustment help
Full change history is on the web site here: Revision History

Next up: log swept sine measurement

Regards,
post #56 of 856
John,

I've been too distracted with work and other projects to spend any more time with Room EQ Wizard, but I wanted to tell you much I like the UI - brimming with all the useful info and very eye-appealing - it makes me feel good just to look at it

And now even better with meters - great!
post #57 of 856
Thread Starter 
Thanks!
post #58 of 856
Thread Starter 
Good news for anyone who has sat through a 10 minute measurement sweep with the Wizard: now it can all be done in about 10 seconds

The default for measurement is now to use a logarithmically swept sine signal, takes 4 seconds for the sweep, about 1.5s of pre and post sampling and, depending on your PC, a few seconds to carry out the FFTs and related processing to generate the responses (my ref for that is a 1.8GHz Pentium M 745 laptop with 1GB RAM). With the extra storage structures needed for the sweep processing the Wizard now peaks at about 55MB RAM usage, so 256MB is the recommended minimum for the PC.

The sweep measurement opens the door for various other interesting features since I now have an impulse response to play with, plenty more to follow

Other changes this time:
  • The graphical meters on the SPL panel looked so nice I added a set to the signal generator panel as well
  • It is now possible to select the generic OS soundcard drivers (on Windows platforms they are Primary Sound Capture Driver, Primary Sound Driver and Java Sound Audio Engine). I do not recommend using these as they typically do not offer the controls the Wizard needs to select inputs and adjust levels, but if the soundcard's own drivers do not fit the bill these offer a fallback. It will be necessary to manually adjust levels via the soundcard's mixer if using these drivers.
  • The control panels now scale up as the app is resized, may help to get better display on other platforms where the fonts don't quite correspond to the Windows versions.
  • The action of the frequency scroll bar has been revised to make it easier to use at small frequency spans
  • Filter optimisation for BFD Pro has been improved
If anyone is concerned that the measurement results for the sweep might not match those with the stepped sine, don't be: they are identical. The sweep has the benefit of being less likely to suffer from clipping when passing through resonances, but rest assured that it still captures the full effect of the resonance and presents the frequency response accordingly. And it is SO much faster...
post #59 of 856
Thread Starter 
There has been some debate about whether room resonances, and in particular the extended decay time they cause, can be fixed with parametric EQ filters. A picture is worth a thousand words, so here is a screen shot I made this afternoon whilst verifying the Wizard's new log swept sine measurements. It shows ETF5 waterfall plots of the subwoofer response in my lounge, with the top plot having a set of correction filters applied via a BFD Pro and the bottom showing the behaviour without any correction. Both plots have the same vertical range, the time axis covers 500ms.



As you can see, without EQ there are some pretty severe resonances between 20 and 30Hz which hang around for quite a while, with another set of problem modes starting at around 47Hz. The EQ filters clean them up very nicely.

For the record, I don't actually use a BFD Pro in my system (my AV processor has parametric filters built in) but I thought it best to use something more people will be familiar with, so spent half an hour or so coming up with some settings for the BFD. The results could be improved further, but I was hungry
post #60 of 856
Thanks for posting those plots John! Very interesting.

Cary
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